Switchvox Administration Reference Manual

Switchvox Administration Reference Manual
Switchvox Administration
Reference Manual
Switchvox SMB version 5.8
©2012-14 Digium, Inc.
All rights reserved. This document is the sole property of Digium, Inc. It contains proprietary
information of Digium. Digium reserves the right to make changes to the specifications described
herein at any time and without notice in order to improve design or reliability of the product. The
information in this document has been carefully checked for accuracy; however, Digium makes no
warranty pertaining to the correctness of this document.
All product and company names are trademarks or registered trademarks of their respective
holders.
630-00003-C
Contents
Setup 6
Extensions
6
Manage (Extensions)
Groups 29
Templates 30
Permissions 31
Settings 31
7
Phones 33
Digium Phones 33
Phone Feature Packs 34
Call Routing
39
Channel Groups 39
VOIP Providers 46
Peered Switchvoxes 56
Outgoing Calls 58
Incoming Calls 60
Admins 63
Manage 63
My Account 64
Tools
65
PBX Features 65
Interactive Voice Response (IVR) Editor
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3
Managing IVR Menus 66
IVR Actions 68
Time Frames 80
Call Recording 81
Digium Addon Products 84
Event Triggers 84
Voicemail 87
Notification 89
SMTP Settings 89
Notification Templates 89
Extension Features
91
Contacts 91
Distinctive Ringtones 92
Switchboard Panels 92
Digium Phone Apps 95
Media 96
Music On Hold 96
Sound Manager 98
Reporting
Call Reports
103
103
Concurrent Calls 103
Call Logs 104
Call Reports 105
Queue Reports
107
Queue Logs 107
Queue Reports 107
Queue Status 108
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Custom Reports
Scheduled Reports
110
110
Server 113
Diagnostics
113
Connection Status 113
Current Calls 115
Server Information 116
Disk Usage 116
SNMP 117
Error Logs 120
Advanced Debugging 120
Technical Support 120
Maintenance 121
Telephony Cards 121
Backups 122
Registration 123
Updates 124
System Clock 125
System Reload 125
Networking 126
IP Configuration 126
Network Proxy Settings 128
Access Control 128
Phone Networks 132
Index 135
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Setup
This section describes the tools under the Setup category. These tools will enable you
to manage your extensions and phones, set up call routing, and give your administrators the appropriate permissions.
Extensions 6
Phones 32
Call Routing 38
Admins 61
Extensions
The Extensions tools give you a means to create and manage Switchvox extensions.
They include the following options:
Manage (Extensions)
7
View Extensions 7
Create an Extension 8
Phone and Virtual Extensions 8
Feature Extensions 16
Feature Code Extensions 25
Bulk-Modify Extensions 27
Import Extensions 27
Groups
29
Manage Extension Groups 29
Extension Group Settings 30
Manage Group Members 30
Templates
30
Permissions
Settings
31
31
System Settings 31
Custom Settings 32
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Manage (Extensions)
When you select Manage, by default, it displays a list of All Extensions in your PBX
system, 50 per page, with their extension number, type, and name of the person
assigned to the extension.
View Extensions
You can filter the list of extensions by selecting the desired Extension Type from the
View dropdown. Options are to display only phones, IVRs, call queues, or virtual
extensions. After a type is selected, the list then displays only extensions of the
selected type.
To find a particular extension, use the Search box, entering at least 3 numbers of the
extension, or at least 3 letters of the first or last name. The list is narrowed to just the
extensions that match what you entered.
To sort a column in the table by ascending or descending order, click the arrow following the column name (Extension, Extension Type, or Name).
Password Warnings
There may be warning icons displayed in the Extension column next to any extension
if the system has evaluated its passwords as weak. Weak passwords can make your
Switchvox vulnerable to intrusion.
By default, Switchvox requires strong passwords be used for extensions and phoneregistrations. These weak passwords may have been created prior to a strong password requirement, or you may have turned that requirement off.
• For details about requiring strong passwords, see “Settings” page 31.
• For details about setting an extension password, see “Numeric Password.”
page 9.
• For details about setting a phone-registration password, see “Phone Password”
page 15.
Actions
Each listed extension has three Action icon buttons:
• Login as this user. Click an extension’s Log In icon to open its My Account page
in the Switchvox Extension Tool Suite without knowing or entering the extension’s password.
NOTE: The extension-owner can be logged in at the same time, so be careful
when either of you are making changes to the extension.
• Modify. Click the extension’s Modify icon to open the Modify SIP Extension
page. See the following section for a description of each of the page’s tabs.
• Delete. Click the extension’s Delete icon to display the Admin Edit Extension
page for deletion. It has two option buttons: Yes, Delete Extension, which deletes
the extension immediately and Download Voicemail/Fax Archive, which saves
and downloads a zip file with the voicemail and faxes associated with the extension.
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Create an Extension
Clicking Create Extension displays the Create Extension/Choose Extension Type
page. Select from the Extension Type dropdown the extension type you are creating.
And if using a template, select it from the Extension Template dropdown. Click the
Create Extension button to display the appropriate following page.
NOTE: If you have not created your own template, leave the template field
as Default.
If creating a SIP Phone or SIP ATA extension, a Profile Information page is displayed.
Complete the profile information, then click Save Extension to save everything and
redisplay the Manage Extensions page, which now lists the new extension.
Extension Types
Switchvox supports many different types of extensions.
See these sections for specific information for each respective extension type:
• “Phone and Virtual Extensions” page 8
• “IVR Extensions” page 16
• “Simple Conference Room Extensions” page 21
• “Meet-me Conference Center Extension” page 21
• “Group Pickup Extensions” page 25
• “Extension” page 23
• “Directory Extensions” page 23
• “Voicemail Access Settings” page 24
• “Call Parking Extensions” page 24
• “Dial Tone Settings” page 25
• “Call Queue Extension” page 17
• “Agent Log In Extensions” page 21
• “Agent Log Out Extensions” page 22
• “Feature Code Extensions” page 25
Phone and Virtual
Extensions
The settings are mostly the same for SIP phones, Analog phones, and Virtual extensions, and each extension type has a corresponding Extension Suite account (/main):
• SIP Phone or SIP Adapter for Analog Phone (ATA). A SIP phone, or an analog
phone plugged into a SIP Adapter (e.g., Sipura SPA-1001).
• Analog Phone. A regular phone (not VoIP/SIP) that is physically plugged into the
Switchvox server. This is an appropriate extension-type for a fax machine or an
intercom.
NOTE: We strongly recommend you do not use this option for your phone extensions. Use instead analog telephone adapters, or a channel bank. If you choose to
use Analog Phone extensions with phones physically plugged into Switchvox,
those phones will not be able to take full advantage of the Switchvox features.
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• Virtual Extension. An extension without a phone or a Switchboard. This type of
extension behaves like a SIP Phone extension except it does not have a Switchboard and it does not show the phone as Unreachable in System Status. This is
useful as a general voicemail box.
Profile Information (tab)
Extension. A unique number for this extension. For information about the extensionnumber length, see “Settings” page 31.
FXS Channel. This is for Analog Phone extensions only. Select the channel that the
phone is plugged into. If the only option here is Unconfigured Channel, then you need
to configure your hardware device channels before this phone can work (see “Channel Groups” page 39).
First Name. The first name of the owner of this extension. This is the name that is
used in the caller ID name, and as the extension’s identification throughout Switchvox.
Last Name. The last name of the owner of this extension. This is the name that is
used in the caller ID name, and as the extension’s identification throughout Switchvox.
Email Address. The email address of the owner of this extension. This is used to
send voicemail-notification messages.
Location. The physical location of the owner of this extension. This is used in the
extension Profile.
Title. The title of the owner of this extension. This is used in the extension Profile.
Language and Locale. The Switchvox web-based tool suite is available in several
languages:
English for United States
English for United Kingdom
Spanish for Spain
Spanish for Mexico
Italian
NOTE: All languages and locales except English for United States handle dates with
the format of dd/mm/yyyy (as opposed to mm/dd/yyyy).
By default, new extensions are created as English for US, with English as the Sound
Prompt Language. If you want new extensions to default to your preferred language,
use “Templates” page 30.
Numeric Password. A Numeric Password for this extension. The password is
numeric because it will be entered from the numeric keypad on a telephone.
The extension owner uses this password to log into their account in the User Tool
Suite. The strength of a new password is indicated as you type (although in this case
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you may be setting a generic password for the owner to log in for the first time). Reenter the password for verification.
You can require that the extension-owner enter a strong password. For details, see
page 32.
Profile Picture
You can upload and modify a picture to include in your profile information.
• Click Upload Picture.
• Click inside the text box to open a dialog box and find the image file on your computer. Images must be at least 75x100 pixels, and must be in jpg format.
• Click Upload.
The image is uploaded and displayed with the crop area that will be used to make
a 75x100 image. You can move the crop area around to choose the best portion of
the picture. The Preview on the right shows you what the picture will look like
after it is cropped.
• Click Upload Picture.
To upload a different picture, re-crop the existing picture, or delete the existing picture:
• Click View Picture.
• Click ReCrop, or Delete. If you want to upload a different image, Delete this
image now, and then click Upload Picture to start again.
• Confirm your changes. Check the appropriate boxes if you want the extension
owner (user) to be able to change his or her Location, Title, or Picture.
Login Actions
Force user to update password on next login. This is a security precaution. It
lets you create all extensions with a generic password (such as the extension
number), which lets the extension owner easily log in for the first time. But then
it forces that person to create a new password before continuing to use the software. (See also page 32.)
Force user to confirm language on next login. If you are not sure which language this extension prefers to use, force the owner to choose a language the next
time he or she logs in. This way, extension owners can decide for themselves
which language to use, and you know that they made a deliberate choice.
Advanced Options
It is unusual to change the Advanced Options, so they are hidden. If you find you need
to change an Advanced Option, click Show Advanced Options.
General Settings
Voicemail/Fax Mailbox Quota (MB). This value is the maximum number of megabytes that can be used for voicemail messages and faxes. It can be set for extensions
and extension groups.
• Blank indicates unlimited space.
• 0 (zero) indicates that no voicemail or faxes can be stored.
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• Quota is set for a group or groups and for an extension in the group indicates
extension's mailbox quota is used.
• Extension is a member of multiple groups, then the lowest mailbox quota is used.
• Extension is not allowed to have voicemail, then setting is not relevant.
Sound Prompt Language. If there are one or more sound packs available in different
languages, those languages are displayed in this dropdown. Select the language that
this extension hears (e.g., when calling the voicemail access extension, the prompts
are in Italian).
To obtain sound packs for different languages, see www.switchvox.com/
product_addons.
For instructions on using sound packs, see “About Sound Packs” page 100.
Minimalist Mode. This mode presents the Web Suite interface with reduced graphics. If Minimalist Mode is set to Yes, the following apply:
• The background graphic and larger graphics on buttons and tabs are removed.
• Some text, especially in section headers, is smaller.
• Animations are removed.
Call API Settings
The call creation API lets external applications originate phone calls via Switchvox.
The API is accessed by requesting a specially formed URL. This request could come
from an external program or by typing the URL into your web browser. To use this
feature, check with your Switchvox administrator.
Phone Settings (tab)
Digium Phone Settings (tab). These settings enable you to control everything about
your Digium phone. To assign a Digium Phone to an extension, see “Digium
Phones” page 33.
Volume levels for the ringer, speaker, handset, and headset are controlled from the
volume bar on the phone itself.
NOTE: The settings are also available in Extension Templates and Bulk Modify, for
your convenience.
General Settings
• Line Label is the label displayed on the phone's Line Key for this extension. You
can enter a specific label, or use variables to describe what the label should be.
Variables are useful in Extension Templates so you do not have to set a unique
label for each extension.
• Label Variables include the following from a dropdown menu: First Name, Last
Name, First Letter of First Name, First Letter of Last Name, Extension.
Msgs Button
• Msgs Button Action is the action that happens when the Msgs button is pressed.
Use Open Visual Voicemail to use the phone Voicemail application to navigate
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via the phone display, or use Dial Voicemail Extension to call the voicemail-system extension (default extension number 899) and navigate via sound prompts.
• Voicemail Extension. If using Dial Voicemail Extension, enter the extension to
dial in the text box.
Idle Screen
• An Idle Screen Image is a PNG image that displays on the phone's idle screen. It
should be 205x85 pixels; but other sizes can be accepted, then scaled.
• Display Missed Calls Notification. If this is YES, a Missed Calls notification is
displayed in the top-left corner of the phone display. If this is changed on the
phone itself using Preferences, then you will no longer be able to change it here in
the web-based suite.
Display
• Brightness and Contrast control the phone display. If this is changed on the
phone itself using Preferences, then you will no longer be able to change it here in
the web-based suite.
• Backlight Dimming. If this is YES, the phone's backlight will dim to the specified
Backlight Dim Level after the phone is not in use for the specified number set in
Seconds until Backlight Dim. If this is changed on the phone itself using Preferences, then you will no longer be able to change it here in the web-based suite.
• Enable Line Keys for Contacts. If YES, the phone's unregistered Line Keys are
used as Rapid Dial Keys.
Sounds
• Default Ringtone is the ringtone used for incoming calls (unless the extensionowner has set up one or more Ring Rules). If this is changed on the phone itself
using Preferences, then you will no longer be able to change it here in the webbased suite.
• Reset Volume Every Call. If YES, the call volume will be reset to the default
after each call. The volume can be changed during the call, using the volume bar.
If this is changed on the phone itself using Preferences, then you will no longer be
able to change it here in the web-based suite.
Answering Calls
• Headset Answer. If YES, when the user presses the Answer softkey during an
incoming call, the call is answered on the headset, not the speaker.This setting
should not be confused with the Auto-answer Switchboard initiated calls feature
under the Common Settings tab. If this is changed on the phone itself using Preferences, then you will no longer be able to change it here in the web-based suite.
• Electronic Hook Switch. Options are Automatic, Plantronics, and Jabra IQ. If
you select Jabra IQ, the Jabra headset must be set to IQ mode. If the Jabra headset does not support Jabra IQ, you may need to update the firmware on the headset. See the manufacturer’s documentation for more information.
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Advanced Phone Options
SIP Settings
• Accept Calls From. If set to This PBX's Networks, then the phone accepts calls
from any of the networks described in Phone Networks. If this is set to Any Host,
then the phone accepts calls from any host.
• Local SIP Port is the phone's port on which SIP traffic is transmitted. You should
not need to change this, but if you do you must be sure that the SIP traffic to and
from the phone is routed correctly.
• Override Dial Plan. If YES, then enter a Dial Plan in the text box.
Passwords
• Phone-admin Password. Enter the phone admin password in the text box.
• Reset Registration for Phone-based Assignment Stealing. No is the default; if
YES, then the phone-registration password for this Digium Phone extension is
reset when its phone assignment changes (meaning, this extension moves to
another phone’s MAC address).And the assignment is changed from the new Digium Phone's numeric keypad (not from the Switchvox web suite, Bulk Import, or
the Extend API). This is referred to as “stealing” a phone assignment because a
phone is taking an extension that already belongs to (is assigned to) another phone.
the assignment is changed from the new Digium Phone's numeric keypad (not
from the Switchvox web suite, Bulk Import, or the Extend API).
Codec Settings
This table displays the codecs your Digium phone supports in priority of use. You can
enable/disable a codec, you can also set its packetization period and jitter settings.
You can move it up and down in the table.
Codec
Bandwidth
Usage (kbits/
s)
Sound Quality
CPU Usage
G722
85
Excellent
Low
G729
30
Good
Medium
G726
54
Good
Low
ULAW
85
Great
None
ALAW
85
Great
None
Other Manufacturers (Tab). To use Phone Settings, your phone must be set up with
a Phone Feature Pack. The Advanced Options are available for all non-Digium
phones.
Common Phone Settings. These can be set for snom phones and Polycom phones.
• Line Label. This is the set of letters and/or numbers that are shown on the line for
your extension. By default, this is your extension number.
• Hide missed calls. If set to NO, this turns off the missed calls notification. You
can turn this back on but you will have to restart your phone, so any calls you have
missed in the meantime will be ignored, and you will not have a missed call notification after the phone is restarted.
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Polycom Phone Settings
These can be set for Polycom phones only.
• Line keys per registration. This sets the number of buttoned-lines on your phone
that are set to this extension. For example, if you have a 3-line phone, you can set
all 3 lines to your extension. That way, you can have an active call and 2 calls on
hold and use the buttons to manage those calls. Any line on the phone that is not
used for your own extension will be automatically filled with your Phonebook
entries. For details, see the Extension-Owner’s User Guide, “Phone Features”
page 13.
• Show caller profile on phone during incoming or during outgoing calls. These
options determine whether or not the Switchvox Profile feature is used on the
phone.
Advanced Phone Options
Config Settings
• DTMF Mode. DTMF tones are the sounds emitted when you press buttons on
your phone. For SIP extensions, refer to your phone's user manual for the DTMF
mode that your phone uses. For SIP providers, ask your provider which DTMF
mode it supports. If you are unsure of which DTMF mode to select, use RFC2833
(the most common method).
Audio Codecs
Select the audio codecs your phone supports. Check your phone’s user manual if you
do not know them.
For audio, ULAW, ALAW, and G722 generally work with every phone and provide
the best sound quality. They are selected by default.
The following table describes the audio codecs. If you have Switchvox SMB and you
have entered a License key for the G729 Codec in Machine Admin > Digium Addon
Products, you will also see G729 in the list of codecs.
Codec
Bandwidth
Usage
(kbits/s)
Sound
Quality
CPU Usage
ULAW
85
Great
None
ALAW
85
Great
None
G722
85
Excellent
Low
G726
54
Good
Low
G729
30
Good
Medium
SPEEX
33
Great
High
GSM
35
Good
Low
ADPCM
54
Good
Low
LPC10
22
Poor
Medium
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Video Codecs
Select the video codecs your phone supports. Check your phone’s user manual if you
do not know them.
For video, H263 and H264 are on by default.
Common Settings (Tab)
General Settings
Auto-answer Switchboard initiated calls. This sets your phone to automatically
answer Switchboard-initiated calls. If you click on a Switchboard phonebook entry,
normally your phone rings and you have to pick it up, then Switchvox dials the number for that entry. If this box is checked, your phone rings and then automatically goes
to Intercom mode and answers the Switchboard, and you hear the phone ringing for
the call to your phonebook entry.
Phone Password. Enter your phone password and verify it. (See also “Settings”
page 31.)
This is the Authentication Password that your SIP phone uses for registration with
Switchvox.
This password may be different than the password used to administer the SIP phone
from the phone itself. Also, this password does not change when this extension's
voicemail password changes.
Phone NAT Traversal. This option controls how Switchvox communicates with the
phone if it is behind a router performing Network Address Translation (NAT). The
following options are available:
• Always. Select if you are unsure if the phone is behind a router performing NAT.
• No. Select if Switchvox is behind NAT but the phone is not.
• Never. Select to prevent attempts to translate even if the handset requests it with
the report tag.
• Route. Select to send responses symmetrically, but not to include the rport tag in
the Via header.
Permissions (tab)
These permissions specify which features the extension is allowed to use. The number
of options set to YES and the total number of YES options are shown in parenthesis
after each permission category. Click the permission’s Arrow button to expand/close
the option and select either YES or NO for the options.
• Change Profile Information. These settings let you control which parts of a profile the extension-owner can change.
• Create Call Rules. These settings let you control which Call Rule actions the
extension-owner can create.
• Use Voicemail Features. These settings let you control which voicemail features
the extension-owner can use.
• Use PBX Features. These settings let you control which general PBX features the
extension-owner can use.
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• Use Digium Phone Apps. These settings let you control which Digium Phone
Apps are on the extension's Digium Phone.
• Change Digium Phone Settings. These settings let you control which of the Digium Phone Settings the extension-owner can change.
• Use Distinctive Ringtones. These settings let you control which Ringtones features the extension-owner can use.
• Use Calling Features. These settings let you control whether the extension-owner
can use these calling features.
• Use Phone Feature Pack Phone Features. This lets you control whether the
extension-owner can use the PBX Applications on a Polycom phone.
• Change Custom Settings. This lets you turn off/on any custom settings.
Outgoing Call Rules (tab)
Specify how this extension can interact with your “Outgoing Call Rules” page 58.
• To allow this extension to use a rule, put the rule in the collection as Allow (green
check mark).
• To deny this extension the use of a rule, put the rule in the collection as Deny (red
forbid sign).
• To ignore an Outgoing Call Rule entirely, do not include the Rule in this collection
(red X).
Extension Groups (tab)
This collector box lets you indicate this extension's membership in Extension Groups.
To put an Extension Group into the Extension Groups collection:
• Type a name into the text field. As you type, Switchvox offers suggestions of
matching groups.
• Click the Find icon to display the groups available in Switchvox.
To delete an item from the collection, select it and click the Delete icon.
To select multiple items in the collection:
• Control-click to select two or more items that aren't together in order.
• Click an item, and then Shift-click another item to select both items and all items
between them.
Feature Extensions
IVR Extensions
This is an extension that presents a menu to the caller, or performs an action. (It is also
known as ‘Auto-Attendant’ in some PBX systems.)
You must have an IVR to create an IVR extension.
For information about setting up IVRs, see “Interactive Voice Response (IVR) Editor” page 65.
The default extension 800 is set with the IVR menu Example IVR. You can customize
that IVR, or create your own.
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IVR Extension
Enter a unique number for this extension. For information about the extension-number length, see “Settings” page 31.
Route to IVR Menu
Select the IVR menu that you want use when this extension is called.
IVR Menu Entry Point
Select an Action that is in the IVR menu. (You don’t have to start at the beginning of
the menu.)
Call Queue Extension
A Call Queue extension lets an incoming call ring a group of extensions so that any
one of those extensions can pick up the call. It displays the Create Call Queue page
with four tabs: Queue Information and Members; Caller Experience; In Queue Call
Routing; and Permissions.
Queue Information and Members (tab)
Queue Setup. The Queue Setup settings determine the basic information about this
queue.
Queue Extension. Enter a unique number for this extension. For information about
the extension-number length, see “Settings” page 31.
Queue Name. Enter the name of the queue (e.g., Sales or Customer Service).
Ringing Strategy. Select one of the following strategies:
• Ring All. Ring all queue members simultaneously until someone answers.
• Round Robin. Take turns ringing each queue member evenly as calls come into
the queue.
• Least Recently Called. Ring the queue member for whom the most time has
passed since taking his last call.
• Fewest Calls. Ring queue member who has taken the fewest calls in the queue.
• Random. Randomly ring queue members.
• In Order. Ring queue members in order, always starting with the first member in
the list.
Queue Members
Queue Members are extensions that can log into the queue and receive calls from the
queue. Only phone extensions are valid queue members. Permanent members are permanently logged in, so they can always receive calls to the queue.
This collector box lets you find and collect extensions or groups that belong in this
queue.
To put an extension or group into the collection:
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• As you type a name or extension number into the text field, Switchvox offers suggestions of matching extensions.
• Find icon displays the extensions available in Switchvox. You can sort this view
by the column headers. If you are working with phone-type extensions, you can
expand the window to show an extension's profile.
• To mark an extension or group as Permanent or Login, select them and click the
appropriate icon.
The order of members in the list is important. For example, if Round Robin is the ring
strategy, it calls the member in the first position, then the second, and so on. If In
Order is the ring strategy, it always tries to call the members in order, starting with the
first member. If you have an extension group in the queue, the group's order is used.
Any non-phone extensions in a group are ignored.
To put items in the right order, select the items you want to move, then click the Up,
Top, Down, or Bottom button.
To select multiple items in the collection:
• Control-click to select two or more items that aren't together in order.
• Click an item, and then Shift-click another item to select both items and all items
between them.
To delete an item from the collection, select it and click the Delete icon.
Queue Member Settings. These settings determine how queue members behave in
the queue.
Seconds to Ring Each Member. This is the maximum number of seconds a queue
member’s phone can ring before the system stops ringing that member and goes to the
next member of the queue.
There are some issues to keep in mind:
• If a queue member has a call rule that answers sooner than this setting, the call rule
is applied. For example, if a queue member has a call rule that sends the call to
voicemail after two rings, and this setting is five, the call goes to that member’s
voicemail and not to the next member of the queue.
• Some phones may stop ringing and give up after some number of seconds or rings.
If this setting is a large value (above 60), and you notice the phones stop before
that number of seconds, you might want to try to change your phone to increase
this limit.
Seconds to Wait Between Members. This is the maximum number of seconds a
queue member’s phone can ring before the system goes on to ring the next member of
the queue.
NOTE: If you set this to a low number and have a large Ring all queue, or a queue in
which no one is logged in, there is potential for system performance issues because
the system keeps trying to find someone to answer the queue call. Unless you have a
specific reason for doing so, leave this value at the default of 5.
Seconds for Wrap Up. This is the minimum amount of time (in seconds) after disconnecting a call before the agent can receive a new call from the queue. For example,
if this is set to 10, an agent’s phone will not ring again for at least 10 seconds after
completing a call.
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Pickup Announcement. This is the selected language and a sound file to play when a
queue member answers a queue call. After this sound is played, the queue member is
connected to the caller. This sound lets the queue member know that the call is from a
queue, and which queue. This is especially useful for people who answer for multiple
queues.
To use the language that has been set for this current call, select Use call language. If
you are only using one language in Switchvox, use this setting. To use a specific language for the pickup announcement for this queue, select that language.
The sound-file dropdown automatically lists all of the sound files in the Sound Manager’s Call Queue folder. For more information, see “Sound Manager” page 98.
Acknowledge Call. Yes indicates members must press 1 to accept the call.
If the member hangs up without pressing 1, then the call goes to the next appropriate
queue member.
Auto Log Off After Missed Calls. This setting controls when members are automatically logged out of this queue. (Permanent queue members can never be logged out.)
• If a member misses more calls in a row than the number in this setting, he or she is
automatically logged out of this queue. Calls will not ring this member until he or
she logs in again.
• If you do not want members to be logged out automatically, set this to 0 (zero).
Receive calls while on a call. Yes indicates that you want this to be true. By default,
queues do not ring members who are already on a call.
NOTE: Your phone must be able to support multiple simultaneous calls for this feature to work properly.
Caller Experience (tab)
The Caller Experience settings offer options for what happens when a caller is waiting
in the queue.
Queue Callers Hear. Indicates whether callers should hear ringing or Music On Hold
while they are waiting in the queue.
If you select Ringing, then Music On Hold and Announcements options are not available.
Music On Hold Group. Indicates the Music On Hold Group you want to play for
callers waiting in this queue.
Announce Position in Queue. If Yes, the caller hears an announcement when they
enter the queue, and the announcement says what the caller’s position is in the queue.
The announcement is repeated depending on the Announcement Frequency setting.
If No, the caller hears Music on Hold with no position announcement.
Announce Estimated Hold Time in Position Announcements. If yes, the caller
hears an estimated hold time along with his position in the queue. The estimated hold
time is derived from how many other calls are in the queue and the average time it
took to answer previous calls.
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Announcement Frequency. This is the number of seconds between announcements.
Manage Queue Announcements
There are various announcement sounds that are played to the caller when he or she is
waiting in a queue. All of these sounds can be customized for this queue. Each
announcement listed here has a description, and a sound-file dropdown to select the
sound that plays when this announcement is triggered.
The sound-file dropdown automatically lists all of the sound files in the Sound Manager’s Call Queue folder. For more information, see “Sound Manager” page 98.
The following selections are available for announcements:
• Language: Plays all announcements from sound files of this language. To use the
language that has been set for this current call, select Use call language. If you are
only using one language in Switchvox, use this setting. To use a specific language
for these announcements in this queue, select that language.
• You are next: The call is now first in line and will be answered by the next available representative.
• Caller Number: This announces to the caller what numeric place he/she has in the
answer queue.
•
•
•
•
Calls waiting: Waiting to speak to a representative.
Hold time: Announces to the caller the current estimated hold time.
Minutes: Minutes
Thank you: Issues this message: Thank you for your patience
In Queue Call Routing (tab)
The In Queue Call Routing settings determine any alternative call routing.
Values entered in the boxes define queue call routing.
• If a caller has been waiting in the queue for [] seconds, route the caller to extension
[], enter a number of seconds, and the extension. If this is left blank, callers remain
in the queue indefinitely.
• If there aren't any members logged into the queue, route all incoming queue calls
to extension [], enter an extension. If this is left blank, callers are kept waiting in
the queue until a queue member logs in.
• If there are [] unanswered calls in the queue, route all new incoming queue calls to
extension [], enter a number and an extension. If this is left blank, there can be any
numbers of callers waiting in the queue.
• If a queue call has passed through the Ringing Strategy [] times without being
answered then route that call to extension [], enter a number and an extension. If
this is left blank, callers remain in the queue indefinitely. The effect of this setting
depends on the Ringing Strategy. For example, if the Ringing Strategy is Ring All,
then the call rings all queue members and that is considered one pass through the
strategy. If the Ringing Strategy is Round Robin, then the call rings each queue
member, and once all queue members have been tried, that is considered one pass
through the strategy.
• While in a queue allow caller to dial a 0 to be routed to extension [], enter an
extension if you want this to be true. Note: Click the People icon to select from a
list of available extensions.
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Permissions (tab)
• Sets the permission level for extensions to display this queue in the Queue Panel
on their Switchboard.
• Adds extensions or extension groups with one of the following permission levels:
My Status: Lets the extension open this queue panel and display his call information for today.
Overview: Lets the extension open this queue panel and display his call information for today, and an overview of all the queue information for today.
Detailed View: Lets the extension open this queue panel and display his call
information for today, an overview of all the queue information for today, and the
members and callers in the queue.
Simple Conference Room Extensions
This extension provides a single conference room for anyone who calls or is transferred to this extension.
Conference Room Extension. This is the unique number for this extension. For
information about the extension-number length, see “Settings” page 31.
Play Sound on Entry/Exit. Yes indicates you want a sound to play as callers enter
and exit the conference.
Play Music On Hold for One Person. Yes indicates Music On Hold will be played
when there is only one person in the conference.
# Out of Conference. This is the extension number to transfer calls to when a caller
presses # in a conference.
Meet-me Conference Center Extension
This extension provides an access point to each user’s conference room. Callers to
this extension are prompted for a PIN corresponding to a specific user’s conference
room.
Extension. This is the unique number for the extension. For information about the
extension-number length, see “Settings” page 31.
Number of Invalid Attempts. This is the number of attempts to allow before transferring the caller to the specified extension.
After invalid attempts route to extension . This is the extension number to transfer
calls to if the caller has entered too many invalid conference room numbers.
Agent Log In Extensions
Agent Log In Settings
An Agent Log In extension is used to log an extension into call queues, and/or to log a
virtual extension into a phone. This extension can also be used to toggle logging in
and logging out, so that you do not have to have a separate Agent Log out extension.
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Extension. This is the unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Require password to login. YES indicates when people call this extension, they are
logged in without having to enter a password.
Login/Logout toggle. Yes indicates members can call this extension to be logged in
or out. If already logged in, callers are logged out. If logged out, they are logged in.
Using this option, you do not need a separate Agent Log Out extension.
Always prompt agents to enter their login extension. YES indicates members must
enter their extension to log in. This is useful if you have people who share work areas
and phones, or if you have virtual extensions that are used by different people.
Affect virtual extension login/logout status. YES indicates this log-in action (or log
out, if toggling) to affect virtual extensions. For example, if a virtual extension is
logged into a phone and into a queue, should the action of logging out apply to the virtual extension, or just the queue?
Queues to Affect
Affect All Queues. YES indicates all of the queues in the system are affected. Otherwise, only the indicated specific queues into the collection are affected.
If you don’t want to affect call queues with this Agent Log In, do not select any
queues.
Agent Log Out Extensions
Agent Log Out Settings
An Agent Log Out extension is used to log an extension out of call queues, and/or to
log a virtual extension out of a phone.
Extension. Enter a unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Require password to log out. NO or YES.
Affect virtual extension login/logout status. NO or YES.
Queues to Affect
Affect All Queues. YES indicates that all of the queues in the system are affected.
Otherwise, enter specific queues into the collection. If you don’t want to affect call
queues with this Agent Log In, do not select any queues.
Intercom / Paging Extensions
An Intercom / Paging extension can make an announcement on a group of speaker
phones. Phones in the group do not ring, they answer immediately in speaker-phone
mode (the caller can begin talking and be heard over the phone’s speaker).
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NOTE: You must have compatible and correctly configured handsets to use this
option.
Intercom /Paging Settings
Extension. Enter a unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Alert-Info header. The value that is sent to the phone in the alert info header. This
setting is required for Polycom phones to auto-answer intercom calls.
Use a value of Intercom or SVAutoPickup. If you use Intercom, a ringtone will play
before the phone auto-answers the call. With both values, the phone auto-answers and
you hear a tone announcing the call.
Timeout. Select the length of time from the dropdown that the system will attempt to
connect an intercom call. After this duration, intercom calls that have not been
answered will be hung up.
Intercom / Paging. Select from the dropdown whether this extension makes intercom
calls (2 Way Intercom) or paging calls (1-way paging).
Choose Sound to Play
Play beep/sound first. Whether or not a beep/sound plays to indicate that the call is
beginning. If you choose YES, then you can also choose a sound to play.
Folder. Select from the dropdown the Folder (within Sound Manager) that contains
the sound that you want to play.
Sound to Play. Select from the dropdown the sound you want to play before you start
talking.
Description . This field automatically displays the script of the sound you selected.
Authorized Extensions and Recipients
Use this section to set up which extensions can use this Intercom/Paging extension,
and which extensions are called when this Intercom/Paging extension is dialed.
Authorized Extensions. Extensions or Extension Groups that can dial this extension.
Recipients. Extensions or Extension Groups that are intercom'd or paged when this
extension is dialed.
Directory Extensions
A Directory extension presents the caller with a directory of all user-type extensions
(extensions with a first and last name). The caller is prompted to type in the first three
letters of the party’s first or last name and is presented with a list of matching extensions.
NOTE: The Directory extension 411 includes the Default Extension Group (which
includes all phone-type extensions).
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Directory Settings
The Directory settings determine how callers can interact with the Directory.
Extension. Enter a unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Press “0” to exit directory and forward to Extension. The extension the caller is
forwarded when he or she presses 0 from the Directory. If this is blank and the caller
presses 0, Switchvox leaves the caller within the Directory.
Search the Directory by members. Select from First Name or Last Name.
Directory Members
Choose the type of members that you want to add, then add items to the collection.
Call Parking Extensions
A Call Parking extension parks a call so it can be retrieved from another phone. Users
transfer the call to the main Call Parking extension, but the call is parked in one of the
extensions in the range specified.
Extension. 700 is preconfigured for a Call Parking extension.
Parking Lot Start and End. The range of extension numbers to be used for parked
calls. This is preconfigured to reserve all extensions between 701 and 799.
For information about the extension-number length, see “Settings” page 31.
Seconds before returning parked calls. The number of seconds that a call should
remain parked before returning the call to the extension that parked the call. The
default is 90 seconds.
Voicemail Access Extension
A Voicemail extension goes directly to the voicemail system where extension owners
can check their voicemail. This is not an extension to leave a voicemail message, it is
strictly for retrieving voicemail. Voicemail extension 899 is preconfigured for you.
Voicemail Access Settings
Extension. Voicemail Access extension is preconfigured at 899. If you want to use a
different extension than 899, first delete the 899 extension, then create a new Voicemail Access extension.
For information about the extension-number length, see “Settings” page 31.
Required password always even when calling from own extension. YES is the
default and indicates a password is always required even from your own extension. If
you do not want to require a password as long as the caller is calling from their own
extension, select NO.
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Group Pickup Extensions
A Group Pickup extension can pick up a call ringing on another extension. For example, assume that the Group Pickup extension is 110, the authorized extension is 111,
and the group includes extension 112 and 113, in that order. If extension 112 is ringing, dialing 110 from extension 111 picks up the call ringing on 112. If extensions 112
and 113 are ringing, extension 112 is picked up (calls are picked up in the same order
as the extensions are listed).
Group Pickup Settings
Extension. Enter a unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Authorized Extensions
Please choose a type . Select either Extension Groups or Extensions to add, and
then put groups or extensions into the collection. A search function is displayed to
assist in finding the desired group or extension.A back arrow redisplays the selection
buttons.
Extensions to Pickup
Please choose a type . Select either Extension Groups or Extensions to add, and
then put groups or extensions into the collection. A search function is displayed to
assist in finding the desired group or extension.A back arrow redisplays the selection
buttons.
Dial Tone Extensions
A Dial Tone extension plays a dial tone when it is called.
Dial Tone Settings
Extension. Enter a unique number for this extension. For information about the
extension-number length, see “Settings” page 31.
Feature Code
Extensions
A Feature Code extension lets you perform a certain operation by dialing an asterisk
(*), followed by a user-defined, two-digit feature code, then another extension. When
you choose one of the Feature Code Extensions, you must always define the two-digit
code to be used by the feature.
Go To Voicemail
This feature directly dials another extension’s voicemail. For example, if you define
*56 as the Go To Voicemail feature code, to transfer a caller to extension 202’s voicemail, dial *56202.
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Directed Pickup
This feature answers a call that is ringing a different extension. For example, if you
define *39 as the Directed Pickup feature code, dialing *39202 will pick up a call
ringing 202’s extension. In addition to defining the code, you choose a type of extension to add, and then put groups or extensions into the collections.
Call Monitor
This feature silently listens in on another extension’s current call. For example, if you
define *92 as the Call Monitor feature code, dialing *92202 will listen in on 202’s call
without either party knowing. While monitoring a call you can switch to different
modes by pressing the corresponding digits on your phone:
• Press 5 for whisper mode. This mode allows you to talk to the extension you're
monitoring but does not let the other party hear you.
• Press 6 for barge mode. This mode allows both parties to hear you.
• Press 4 to return to the default monitor mode.
Choose a type of extension to add, and then put groups or extensions into the collections.
Personal Intercom
This feature directly dials another extension’s intercom. For example, if you define
*55 as the intercom feature code, and you dial *55202, you will now be talking over
the speaker on extension 202’s phone.
NOTE: This feature requires intercom support on the handset receiving the call.
Personal Intercom Settings. In addition to the two-digit feature code, enter the
Alert-Info Header, select the Timeout timing and Intercom/Paging from the dropdown menus.
Choose Sound to Play. Enter the following intercom information:
•
•
•
•
Play Beep/Sound First. YES or NO.
Folder. Select from the dropdown menu the folder in which to store the sound.
Sound to Play. Select the sound from the dropdown menu.
Description. A text description of the selected sound is automatically displayed.
Personal Intercom Groups. Choose a type of extension to add, and then put groups
or extensions into the collections. At least one group must be able to use this feature
code.
Fax
This feature sets a system extension as the sender of the fax, and then sends a fax. The
fax is stored in the Sent Faxes of the designated extension. Select YES from Require
Extension Password if you want to require the sender to enter the password for the
designated extension.
For example: you have a Fax Feature Code extension of 33. You want to send a fax as
if it were from extension 200, which has a voicemail password of 111. You must dial 9
for an outgoing line, and you are sending the fax to 1-888-555-1212.
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You would dial the following from your fax machine:
*33200*111*918885551212. If the password is not required, leave it out:
*33200*918885551212. In both cases, the fax is sent to 1-888-555-1212 and is
stored in the Sent Faxes folder of extension 200.
Bulk-Modify
Extensions
Import Extensions
The Bulk-Modify Extension tool lets you search for and select one or more extensions, then modify specific settings for them at the same time. Under Phone Extensions, type some search criteria, and the system will display extensions meeting your
criteria.
1.
Select the extensions you want to modify, and they are added to the Extensions
collection box.
2.
Under Phone-Extension Fields, select the type of setting you want to modify.
Options are Profile Information, Phone Settings, Permissions, Outgoing Call
Rules, and Extension Groups.
3.
Then select a setting within the selected type.
4.
Specify the value.
5.
Click Add or Cancel.
6.
Review the extensions and the changes that will be made to those extensions.
7.
Click Save Modifications.
The Import Extensions tool lets you create and update multiple SIP or Virtual extensions from data in an imported CSV file.
The Import Extensions File
The Extension File must be in Comma Separated Value (CSV) format.
The first line of the file must be a header line that labels each column. Each subsequent line should be an extension line that contains the values of an extension.
Depending on the type of extensions you are working with (SIP or Virtual) different
information is required or accepted. You can use a file that has additional columns of
information (Switchvox ignores this information) as long as the required extension
information is included correctly, The order of the labels in the header is not important, but the values of each extension must match the order of the labels.
If you are working with SIP extensions, the value of phone_password cannot be the
same as extension.
If you are updating an existing extension and you have some information that you do
not want to change, use just the comma with no spaces or text. If you want to make it
blank, put a space in that column.
When your Extension File is ready, go to Setup > Extensions > Manage and click
Import Extensions (see “Import Options” page 28).
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File Information
This table describes the data for each type of import:
ext
fname
lname
email
title
location
password
phone_
password
mac_
address
SIP:
New
X
X
X
X
A
A
X
X
A
SIP:
Update
X
A
A
A
A
A
A
A
A
Virtual:
New
X
X
X
X
A
A
X
NA
NA
Virtual:
Update
X
A
A
A
A
A
A
NA
NA
X = Required :::A = Accepted ::: NA = Not Used
File Examples
SIP Extension File
ext,fname,lname,email,password,phone_password,mac_address
100,Joe,Smith,[email protected],100,[email protected]$,0019159C0646
101,Jane,Doe,[email protected],101,[email protected],0013174C5532
102,Dan,Jackson,[email protected],102,[email protected]$$,
Virtual Extension File with location and other information
fname,lname,email,dept,ext,password,location
Joe,Smith,[email protected],sales,100,100,3A
Jane,Doe,[email protected],sales,101,101,3B
Dan,Jackson,[email protected],facilities,102,102,9A
SIP Extension Updates File without passwords with a location
ext,fname,lname,email,password,phone_password,location
100,Joe,Smith,[email protected],,,2A
101,Jane,Doe,[email protected],,,3B
102,Dan,Jackson,[email protected],,,2C
Import Options
The Import Extensions options determine how to manage this import. When the settings are correct, click Evaluate Import to evaluate the Extension File and see what
will happen if you import the file.
Extension File. Browse your machine to find your Extension File.
For details about creating this file, see “Import Extensions” page 27.
Extension Type. This is the extension type that is used for all of the extensions that
are created.
Extension Template. This is the Extension Template that is used to create or update
all of the extensions. All of the values in the template are used for each of the extensions in the Extension File.
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On Duplicate Extension. This option specifies how to respond when an extension in
the Extension File already exists on the system.
• Skip: Do nothing with the extension data in the file, and skip to the next extension
in the file.
• Delete and Recreate: Delete the existing extension in the system, then create the
extension with the values from the file and the chosen extension template. Note:
only SIP, Analog and Virtual extensions are affected by this. For example, if you
have a Call Queue extension of 333 and have a line in your file to create an extension of 333, this line in the file is skipped even if you have Delete and Recreate
chosen.
• Update Information: Update the existing extension with the values from the file
and the chosen extension template. NOTE: the update is only applied if the extension types are the same. For example, if you have an existing Virtual extension of
100 and have a line in your file to create a SIP extension of 100, this line in the file
is skipped even if you have Update Information chosen.
Import Extensions Evaluation
When you click Evaluate Import, Switchvox opens your Extension File and evaluates
it against the extensions that already exist.
The results are shown with how many extensions will be created, how many are duplicates, what will be done with those duplicates, and how many are invalid and cannot
be created.
If you have invalid extensions, check your Extension File for formatting problems or
errors. Be sure your extension numbers are the same length as specified in Setup >
Extensions > Settings.
If you want to proceed with the import, click Yes, Process Extensions.
Groups
The Extension Groups tools enable you to group extensions together so you can apply
changes and features to several extensions at a time.
NOTE: The All Extensions group includes every extension in Switchvox. If you
add an extension, it is automatically included as a Group Member. This group
cannot be modified.
Manage Extension
Groups
Create Extension Group adds a new extension group.
The Modify icon for the group lets you select the group you want to modify and display the Modify Group page.
The Delete icon for the group selects the group you want to delete and displays the
Delete Group page. Click Yes, Delete Extension Group to remove the group.
Hover over the Note icon to see additional information added for a Group.
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Extension Group
Settings
Name. A unique name for this group.
Note. A descriptive note for this item. This is displayed when you hover over the Note
icon.
Voicemail/Fax Mailbox Quota (MB). A number of megabytes as a disk-space quota.
This is the maximum number of megabytes that can be used for voicemail messages
and faxes. This value can be set for extensions and extension groups.
• Leaving blank, indicates unlimited space.
• Entering a 0 (zero), indicates no voicemail or faxes can be stored.
• If a quota is set for a group or groups, and for an extension in the group, the extension's mailbox quota is used.
• If an extension is a member of multiple groups, the lowest mailbox quota is used.
• If an extension is not allowed to have voicemail, this setting is not relevant.
Manage Group
Members
This collector box lets you find and collect extensions and other Extension Groups
that belong in this Extension Group.
To put an extension or Extension Group into the Group Members collection:
• Type a name or extension number into the text field. As you type, Switchvox
offers suggestions of matching extensions.
• Click the Find icon to display the extensions available in Switchvox. You can sort
this view by the column headers. If you are working with phone-type extensions,
you can expand the window to show an extension's profile.
To delete an item from the collection, select it and click the Delete icon.
To put items in the right order, select the items you want to move, then click the Up,
Top, Down, or Bottom button.
To select multiple items in the collection:
• Control-click to select two or more items that aren't together in order.
• Click an item, and then Shift-click another item to select both items and all items
between them.
Templates
Extension Templates contain almost all of the same settings as the Phone and Virtual
Extensions, except for the personal information of the extension-owner.
Templates even contain the membership in Extension Groups, so using well-defined
Templates and Groups can make it simple to set up new employees.
If most of your phone-type extensions will be set up the same way, you can modify
the default template to suit your needs. If not, you can create multiple templates for
different types of employees.
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Each time you create a SIP, Virtual, or Analog extension, you choose an Extension
Template. Then, when the Create Extension page opens, all of the settings match that
Template, and you don't have to do all of that work again.
Extension settings can always be changed. If you have a special case, you can create
the extension with a template, and then modify the extension to change one or two settings.
Permissions
The Extension Permissions page displays the existing extension permission mappings
in a table that includes extension holders, their respective target extensions, and the
associated permissions. You can modify the information or delete the mapping
entirely.
If you need to create a permission mapping, click the Create Extension Permissions
Mapping button. This displays the Create Extension Permissions Mapping page. On
this page, you choose the type (extension group or individual extension) who will be
the Permission Holder and the type (extension group or individual extension) who
will be the Permission Target. Then you specify the permissions. The following are
options:
• Permission holders can view target’s calls. YES/NO let the holders view the
caller ID of the target’s current calls.
• Permission holders can record target’s calls. YES/NO let the holders record the
target’s current calls.
• Permission holders can monitor target’s calls. YES/NO let the holders silently
listen in on the target’s current call. This includes monitor, whisper, or barge.
• Permission holders can pick up target’s ringing calls. YES/NO let the holders
answer a call that is ringing the target's extension.
• Permission holders can use intercom to contact target. YES/NO let the holders
initiate an intercom connection with the target extension.
When finished, click Save Extension Permission Mapping. The Extension Permission page is re-displayed showing the new mapping.
Settings
The Extension Settings page allows you to set the default System Settings as well as
to define Custom Settings for a phone extension. Click Save Extension Settings
when finished.
System Settings
General Settings
• Extension Length. Select 3, 4, 5, or 6 digits. This is the length of the extensions in
Switchvox. Changing this number does not change existing extensions.
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NOTE: If you specify an extension length that is longer than some existing extensions, there may be a slight delay in internal dialing. For example, if you call a 3digit extension, Switchvox waits a moment before connecting the call, to see if
you are going to dial a fourth digit (or fifth or sixth).
• Beep after an assisted transfer. YES indicates you want to play a beep sound
when an assisted transfer is done. For example: Jill puts Jack on hold then calls
Jane, and asks her if she can speak with Jack. Jane agrees, and Jill transfers Jack to
Jane. When the transfer is complete the beep is played, Jill is off the call, and Jack
and Jane are on the call and can begin talking.
Security Settings
• Require strong extension passwords. YES indicates you require extension-owners to use strong extension passwords, and the extension-owner will not be able to
change his or her password unless it is evaluated as strong. A strong password has
at least 6 numbers, no more than 2 of the same number in a row (e.g., 111), and no
sequences (e.g., 123). Also, it does not contain the extension.
For details about the extension setting that this affects, see “Profile Information
(tab)” page 9
• Require all extension owners to change their password if it is not already strong.
YES indicates that if strong passwords are required, then you also require that all
extension-owners change their password if it is not already strong. This means
extension-owners cannot use the Web Suite until they have created a strong password. This is similar to Login Actions, but this setting applies to all extensionowners who do not have a strong, or very strong password.
• Require strong phone registration passwords. YES indicates you require phones
register with Switchvox using a strong password. A strong password has at least
six characters, one of each character type (uppercase letter, lowercase letter, number, and symbol), no sequences (abc, 123), no more than 2 of the same character
type in a row, and includes 2 symbols. For example: m!2T^1
If you do not want to use symbols, you can just make the password longer. For
details about the extension setting that this affects, see “Phone Password”
page 15.
NOTE: If you are using Digium Phones, or you used Phone Feature Packs to configure a phone, then the phone is already configured with a very strong password.
Advanced Settings
Dial Plan Settings
Use 10-digit local dialing. NO/YES.
Max Extension Length Match. Select length from the dropdown menu.
Custom Settings
Custom Extension Settings are used to pass extension-specific values to uploaded
phone apps. The following can be set:
• Name. Name of the custom setting.
• System Key. Automatically copied from the Name value.
• Value is a Password. Indicate YES/NO if the Name is the Password.
Click Save Custom Setting when finished.
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Phones
The Phones tool allows you to select either Digium Phones or Phone Feature Packs to
make phone assignments.
Digium Phones
33
Digiium Phone Assignments 33
Quick-Create Assignments 34
Phone Assignment Options 34
Phone Feature Packs
35
Unknown Phones (tab) 35
Unconfigured Phones (tab) 36
Configured Phones (tab) 36
Making Phones Visible to Switchvox 37
Phone Setup Options 38
Digium Phones
When you select this option, it displays the Digium Phone Assignments page, which
lists the unassigned extensions by default. There are two more pages: Quick-create
Assignments and Phone Assignment Options.
Digiium Phone
Assignments
Phone assignments define the Switchvox SIP-Phone extension assigned to a Digium
Phone's MAC address. An assignment must define Line Key 1 as a Switchvox SIPPhone extension, but an assignment can also define lines 2-6. Those lines can be
another SIP phone extension on Switchvox or an external extension.
You can filter the phone assignments for viewing from among the following types:
• Unassigned Phones. Digium Phones on the network that do not already have a
Switchvox extension assigned to them.
• Unassigned Extensions.Switchvox extensions that do not already have a phone (a
Digium Phone or otherwise).
• Assigned Phones and Extensions. Displays the following:
-MAC Address for the Digium Phone
-Model number for the Digium Phone
-Configured By indicates if the assignment was created by the Server (the
Switchvox Web Suite, Bulk Import, or the Extend API), or by the Phone itself
(using the numeric keypad to select the extension for Line Key 1). Server-based
assignments cannot be changed from the phone itself.
-User Details indicate the extensions assigned to the phone. Line 1 must be a
Switchvox phone extension, but lines 2-6 (depending on the model of your Digium Phone) can be a Switchvox extension, or an external extension.
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• Ignored Phones and Extensions. Digium Phones or Switchvox extensions that
you do not want to assign and you do not want to see in the Unassigned lists. If
you want to delete a Digium Phone entirely from Phone Assignments, first ignore
it, then you can delete it.
Quick-Create
Assignments
The Quick-Create Assignments button lets you assign multiple phones and extensions
quickly. Using this option, the extension is always assigned to Line 1.
This is most useful if you have a barcode scanner, so that you can scan each phone
box, and easily associate it with the correct extension.
The Populate Unassigned Extensions button allows you to add all unassigned phone
extensions as rows in order by extension number, first name, or last name.
Click Save All Assignments when finished.
Phone Assignment
Options
These options can be used for creating phone-based assignments using the phone's
numeric keypad to select an extension from a list. You don't need to use these options
if you create server-based assignments (meaning, assignments created from the
Switchvox Admin Suite, Bulk Import, or the Extend API).
Server Name. This is the name of the Switchvox server as it appears in the phone's
list of Digium Configuration Servers.
Assignment Authentication. To ensure that your phones are assigned correctly, you
can choose a method of authentication to be used before the phone will be assigned:
• Assignment Code: One code displays the extension list and lets any extension be
assigned.
• Extension / Password: The extension list is always displayed, but after an extension is chosen, that extension's password must be entered.
Assignment Code. Enter a numeric code that enables you to assign extensions.
Enable Open Enrollment Period. This option sets a period of time where phonebased assignments can be created without any authorization. Use this option with caution.
Click Save Assignment Options when finished defining or making changes.
Phone Feature Packs
The following must be in place before you use Phone Feature Packs to set up your
phones:
• Switchvox is up and running.
• Your phones are on the same network as Switchvox.
• You have a Phone Feature Pack available for each phone.
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• Any Polycom phones you are setting up are running SIP firmware 1.6.7 or higher.
• Any snom phones you are setting up are running snom firmware version 6.5.18 or
higher.
• You are comfortable with Switchvox updating each phone’s firmware as necessary. If you do not want a phone’s firmware updated, you need to set up that phone
manually (not as part of this process).
• You are comfortable with Switchvox setting each phone’s config authentication
login and password, and its handset password. The config authentication login and
password are randomly generated and can be found in the Configured Phones list.
The handset password is always 456. If you do not want this information changed
for a phone, you need to set up that phone manually (not as part of this process).
IMPORTANT: If your phone is not on the same network as Switchvox, or it does
not have the required firmware, it cannot be discovered by Switchvox. If this is the
case, see “Making Phones Visible to Switchvox” page 37. If you have a Polycom
300 or 500 phone, you must set up your phone manually (you cannot use the
Phones tool). For help, see the documentation for your phone.
The Phones tool has three main tabs: Unknown Phones, Unconfigured Phones, and
Configured Phones. You must confirm each phone for configuration, then configure
each phone with extension details.
To begin setting up phones on Switchvox, do the following:
• Place each phone where it should be located, plug it into your network, and turn it
on. It is helpful to note the MAC address of each phone, but it is not necessary.
When each phone boots up (after you turn it on), Switchvox detects it and adds it
to the Unknown Phones list. These are phones that Switchvox has seen on the network, but does not have control over.
• In the Admin Suite, select Setup > Phones. > Phone Feature Packs
The Phone Feature Packs page opens to the Unknown Phones tab. If a phone does
not appear in the list, try rebooting the phone.
Unknown Phones
(tab)
Unknown phones are the phones that Switchvox has found on your network, but they
are not yet configured for use with Switchvox. (These phones do not receive any configuration files from Switchvox.)
Phones that have not been detected on the network for more than 14 days will stop
showing up in the Unknown Phones list.
• Check the box next to each phone you want to configure, then click Confirm
Checked Phones. Switchvox sends firmware and configuration files to the phone,
then a command to reboot.
IMPORTANT: If you do not want to configure a phone, do not check its box.
That ensures that the phone does not receive Switchvox configuration files. In particular, if you have more than one PBX on your local network, make sure that you
do not confirm phones that are set up on another PBX. This overwrites the configuration of these phones and they will no longer connect properly to the other PBX
Phones.
• Be sure that each phone has finished rebooting. This can take as much as 10 minutes, and it is important that it is finished before you move on to the next step. If
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the date and time are shown on the phone’s display, that indicates that it has finished rebooting.
After your phone has finished rebooting and is back online, it is in the Unconfigured Phones list. Now it needs to be assigned an extension.
Phone Model Not Reported
Switchvox is not able to determine the phone model for some phones. Despite that,
these phones can still be configured. During the configuration process, Switchvox
updates the firmware on the phone (if necessary), and that helps to determine the
model.
Unconfigured Phones
(tab)
Unconfigured phones are the phones that are registered with Switchvox and can
receive configuration files, but these phones do not yet have an assigned extension.
NOTE: The ‘Config Authentication’ Login and Password shown on this page are
used when the phone requests a current configuration file from Switchvox. This
is different from the phone’s registration login and the extension’s login, and you
can safely ignore this information.
To remove a phone from this list and put it back on the Unknown Phones list, check
the box for that phone and click Reject Checked Phones.
To assign an extension to a phone:
• Click the Modify icon
• Enter an extension number, or search for the extension that you want.
• After each phone has the correct extension, check its box and click Confirm
Checked Phones.
Switchvox pushes out the final configuration to each phone, and sends a command to
each phone to reboot again.
After your phone has finished rebooting and is back online, it is in the Configured
Phones list. If you want to assign additional lines to this phone, you can do that on the
Configured Phones page.
Configured Phones
(tab)
Any phones you configured are now displayed under the Configured Phones tab on
the Phone Feature Packs page. Each time a configured phone reboots, it receives configuration updates and its user’s specified Phonebook (see the Extension-Owner User
Guide, “Phone Features” page 13).
To modify a configured phone, click its Modify icon.
Main Line. Set an extension and an alternate host for this phone.
Additional Lines. To add a line, enable it, then specify whether this is a Switchvox
Extension or not, and then enter the extension information.
• If it is a Switchvox extension, you just need to enter that extension number.
• If it is not a Switchvox extension, you can enter host name, alternate host name (if
available), Port, User ID, and Password.
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Making Phones
Visible to Switchvox
If Switchvox cannot discover your phone (i.e., you cannot see your phone in the
Unknown Phones list), you need to manually point the phone to the Switchvox server.
The following describes the process for Polycom and snom phones:
Polycom Phones
• Reset the phone back to its factory defaults:
For Polycom models 320, 330, and 430: hold down the 1, 3, 5, and 7 keys simultaneously.
For other models: hold down the 4, 6, 8, and * keys until the phone prompts you
for a password. Then press 4, 5, 6 on the keypad, then select Enter. Once the phone
reboots, press Setup during the countdown screen. Then press 4, 5, 6, on the keypad, then select Ok. Go into Server Menu to find the settings.
The phone reboots with the factory settings.
• Go to the Network Configuration for your phone: press Menu, 3,2, 456 (the password), Enter,1, 1. Scroll to Server and select it.
• Set server type to http and server address to IP_ADDRESS_OF_YOUR_PBX/pc.
For example, if your PBX has an external IP address of 192.100.0.1 then set the
server address to 192.100.0.1/pc.
snom Phones
Reset the phone back to its factory defaults.
Set the Setting URL to ‘http://SWITCHVOX_IP_ADDRESS/pc/{mac}.xml.
For example, if Switchvox's IP address is 192.100.0.1, the Setting URL is http://
192.100.0.1/pc/{mac}.xml.
After those changes, make sure that the phone is rebooted. Rebooting can take as
much as 10 minutes, and it is important that you let it finish completely. If the date
and time are shown on your phone’s display, that indicates that your phone has finished rebooting.
When your phone reboots and comes back online, it appears in the Unconfigured
Phones list in Phone Setup.
IMPORTANT: To configure phones that are external to your network, Switchvox must be accessible outside of your network over port 80. Please contact your
network administrator for information about how to do this.
Phone Setup Options
Fromthe Phone Feature Packs page, the Modify Phone Setup Options button allows
you to modify the following setup options:
Alternate Host Default Option. If Alternate Host is not set, use the PBX’s external
IP/hostname. YES indicates the phone uses the default for the Alternate Host option.
The default is only applied if a phone does not have a setting in the Alternate Host
option.
Security Option. Only accept calls from the PBX. YES indicates your Switchvoxconfigured phones should only accept calls from this Switchvox.
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Digit Map Options. If you wish to specify a custom digitmap to use with your
phones, you may do so with the following options. (Leave them blank if you are
unsure.)
• Polycom Digitmap
• Polycom Digitmap Timeout
• Snom Digitmap
Otherwise, when your phone gets configuration files from Switchvox, then Switchvox
provides digit map settings for your phone based on the extension lengths your system
uses and typical dialing requirements.
Phone Discovery Options: DHCP Sniping. Polycom phones with SIP Applications
prior to 2.2.2 do not support DHCP INFORM. To configure these older phones,
Switchvox must be more aggressive, usurping your network’s DHCP server. After the
phone is upgraded, Switchvox returns to normal behavior.
It’s probably safe to leave this option enabled, but if your DHCP server reacts poorly
when older phones are configured, disabling this option might help.
NTP Options. Custom NTP Server. You can specify an Network Time Protocol
Server to set the time on your Switchvox-configured phones.
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Call Routing
Tools are available for managing the following call routing options:
Channel Groups
39
Device Types 40
Callback Extension 41
Default Fax Extension 41
Advanced Options 41
VOIP Providers
46
SIP Providers 46
IAX Providers 52
RSA Key 55
RTP Port Range 56
Peered Switchvoxes
56
Configure New Peer 56
View Currently Configured Peers 57
Local Peer Settings 58
Outgoing Calls
58
Outgoing Call Rules 58
Caller ID Rules Tab 59
Call Diagnostics Tab 60
Incoming Calls
60
DID Routes Tab 60
Caller ID Rules Tab 62
Channel Groups
If you have analog or digital lines you want to connect to Switchvox, you need to purchase the appropriate Digium Telephony Card, and set it up using the Channel
Groups tools. These tools let you manage hardware device channels to enable multiple signaling methods and to configure callback extensions. To create a channel
group, click Create Channel Group. This displays the Channel Group page.
•
•
•
•
Group Name. Enter a name for the channel group.
Device Type. Specify the device type. from the dropdown menu.
Add Channels to this Group. Select channels to add to the group from the list.
Callback and Default Fax Extension. If applicable, enter the Callback Extension
and the Default Fax Extension for the channel. Voice calls that do not match any
Incoming Call rules or routes are sent to the Callback Extension. If you have fax
installed, Fax calls that do not match any Incoming Call Routes are sent to the
Default Fax Extension.
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NOTE: Unless you are an advanced user, do not change any of the other settings.
When you are done, click Save Channel Group to create the group.
WARNING: Switchvox restarts after you create a new channel. Any current calls
are dropped, and any agents logged into queues are logged out.
Device Types
Device Type describes the type of line that is connected to the PBX through this
device. The following options are available:
FXO and FXS. Foreign Exchange Office (FXO) and Foreign Exchange Service
(FXS) device types refer to the type of interface on the telephone equipment: an FXO
interface receives power (battery) and receives ring signals; and an FXS interface provides power (battery) and generates ring signals. FXO devices can only communicate
with FXS devices and vice versa. Each FXO and FXS type listed in the dropdown has
an associated signaling type: kewlstart, loopstart, or groundstart. The signaling type is
the “language” the interface uses to communicate to the devices connected to it. Some
devices have their signaling type preset and cannot be changed (for example, analog
cards). Some devices can have their signaling type changed to match the devices they
are connected to.
NOTE: You cannot have both FXS and FXO channels in the same group. However, you can create more than one group per signaling type.
FXO (Foreign Exchange Office) with Kewlstart signaling. Select this option if you
are connecting normal analog phone lines from the phone company into your PBX.
This option can also be used with certain T1 lines or T1 channel banks in rare
instances.
FXO (Foreign Exchange Office) with Loopstart signaling. Do not use this option.
Use FXO with Kewlstart instead. Kewlstart is fully compatible with Loopstart, and
will detect hangups on most lines which the Loopstart option does not do.
FXO (Foreign Exchange Office) with Groundstart signaling. Groundstart is an
alternate type of signalling used with analog phone lines from the phone company.
Groundstart phone lines are slightly unusual, so double-check with your provider
before selecting this option.
FXS (Foreign Exchange Station) with Kewlstart signaling. Select this option if
you are connecting a normal analog handset to an FXS port of an analog card.
FXS (Foreign Exchange Station) with Loopstart signaling. It is recommended to
use the Kewlstart option instead of Loopstart.
FXS (Foreign Exchange Station) with Groundstart signaling. Select this option if
you are connecting a groundstart FXS device, such as another analog PBX, to the
FXS port of an analog card.
PRI. The Data channel is a single channel used for call control on T1/E1 PRI connections. It is usually on channel 24 for T1, and 16 for E1 connections. The Bearer channels carry the audio for T1/E1 PRI calls. Full PRIs use all channels except for the D
channel. Partial PRIs use a subset for the Bearer channels.
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PRI T1/E1 D (Data) Channel. The Data channel is a single channel used for call
control on T1/E1 PRI connections. It is usually on channel 24 for T1, and 16 for E1
connections.
PRI T1/E1 B (Bearer) Channel. The Bearer channels carry the audio for T1/E1 PRI
calls. Full PRIs use all channels except for the D channel. Partial PRIs use a subset for
the Bearer channels.
BRI. The Data channel is the third channel on every BRI that controls signaling. The
Bearer channels are the first two channels on a BRI that actually pass calls.
BRI D (Data) Channel. The Data channel is the third channel on every BRI that controls signaling.
BRI B (Bearer) Channel. B channels are the first two channels on a BRI that actually pass calls.
E&M Signaling. The E&M (Ear & Mouth) Signaling device type is a different type
of T1 signaling used by some telephone companies. E&M signalling describes certain
older T1 lines. Select this option if you have an E&M or E&M-wink circuit, sometimes referred to as a Supertrunk.
Callback Extension
This is the extension to ring when a voice call comes in over this provider or channel.
Callback extensions can be any extension type (your receptionist’s phone, an IVR,
etc.).
Default Fax Extension
This is the extension to ring when a call comes in over this provider or channel, and is
identified as a fax. (Switchvox only identifies T.38 faxes over SIP providers. A fax
coming in over an IAX provider or a channel, or a fax using a different standard may
not be identified as a fax.)
Advanced Options
It is unusual to change the Advanced Options, so they are hidden. If you find that you
need to change a setting, click Show Advanced Options.
Aggressive Echo Cancellation. This option enables aggressive echo suppression in
the Analog echo canceller. This is done by making the line half-duplex, or by muting
the party that is not actively speaking. While this does eliminate echo, it might cause
other problems such as not being able to interrupt a speaking party, or clipping on the
leading and trailing edge of speech. Setting this option to run for the first 5 seconds of
a call is an effective way to eliminate echo that eventually trains out. Otherwise, it's
best to keep this option disabled.
Apply Noise Reduction Filter. This option applies a filter on calls to reduce analog
line noise. This can improve quality on very old analog lines in rural areas. Before
enabling this option, it's best to call your provider to see if they can improve the signal
quality or strength from their end, as this option only masks over the noise.
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Automatic Gain Correction. Enabling this will automatically adjust the received
volume of calls to a standard volume level. This should be the first thing to try if
inbound calls are too quiet, specifically on analog lines.
Automatically determine call progress. Do not enable this option unless explicitly
instructed to. Enabling this option in the wrong environment will result in dropped
calls.
Battery Debounce. This option is extremely advanced and only used in very rare circumstances in certain countries.
This option controls how Switchvox looks for a hangup signal on Kewlstart lines. In
Kewlstart signalling, hangup is sent by a voltage drop in the analog lines for a certain
period of time.
The Battery Debounce controls how longSwitchvox waits until concluding that it has
received a hangup signal. If set too low, line noise or lightning or regular speech can
be interpreted as a hangup. If set too high, correct hangup signals will not be received.
The default value for this setting is chosen to be the perfect balance. Do not change it
unless you've physically measured the duration of a battery drop for your country's
telephone lines.
Battery Threshold. This option is extremely advanced and only used in very rare circumstances in certain countries.
This option controls how Switchvox looks for a hangup signal on Kewlstart lines. In
Kewlstart signalling, hangup is sent by a voltage drop in the analog lines for a certain
period of time.
The Battery Threshold controls how low the voltage must go before being considered
dropped. The default setting for this option will work in most countries. Do not
change it unless you've physically measured the duration of a battery drop for your
country's telephone lines.
Caller ID Signaling. Caller-ID is sent over analog lines differently in different countries. This option controls which method to read the caller-ID. The options are Bell
(for countries like the USA), v23 (United Kingdom), and DTMF (Sweden). When
changing this value, also make sure to adjust the Caller ID Start value to correspond to
your country.
Caller ID Start. Different countries send caller-ID at different times. This option
instructs Switchvox when to look for the caller-ID signal. The choices are ring (for
countries like the USA), or polarity (for countries like the United Kingdom).
Detect busy signal. This option will listen for the busy signal to trigger a hangup. It is
strongly recommended to leave this option disabled and have your telco send a true
disconnect signal. This option may be enabled in countries that do not have disconnect supervision. If you are in the USA, UK, Australia, or Canada, you should not
need to enable this option. If this option is enabled, it is advised to increase the Busy
Count setting to as many tones as possible to avoid false-positives that may result in
dropped calls.
Echo Cancel. This option enables or disables echo cancellation on the system. We
strongly recommend setting this option to Yes. Setting this option to No will disable
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both software echo cancellation, and any onboard hardware echo canceller on the
given card.
Echo Cancel when bridging TDM channels. This option controls echo cancellation
for calls that enter Switchvox on one hardware channel, and exits Switchvox on
another hardware channel. IE: this affects calls that do not touch any VoIP Providers
or SIP handsets.
Calls of this nature do not normally need echo cancellation, so it is safe to leave this
option to the default of No.
NOTE: This setting does not apply to calls to normal SIP handsets.
Echo Canceller Tail Length. This controls how long the canceller should listen for
echo to return. It is best to use the smallest value possible. For analog lines, echo is
usually in the 64 or 128 range. For T1 lines, 128 or greater.
Echo Canceller Type. Different echo canceller types may perform better in certain
situations.
Fax Buffer Policy. Set the Fax Buffer Policy for this Channel Group. To turn this
option off, select Disabled. The default is Half. This option lets you set a buffer policy
to prevent fax failures on high latency systems. See the related option “Number of
Fax Buffers” page 44.
NOTE: You must install a fax license and the fax software to make faxing available. For details, see “Digium Addon Products” page 84.
Fax Detection. Set the Fax Detection for this Channel Group. For more information,
see About Faxing.
Both. Detect incoming and outgoing faxes, and behave appropriately.
No. Do not detect incoming or outgoing faxes.
Incoming. Detect incoming faxes and behave appropriately. Do not detect outgoing faxes.
Outgoing. Detect outgoing faxes and behave appropriately. Do not detect incoming faxes.
NOTE: You must install a fax license and the fax software to make faxing available. For details, see “Digium Addon Products” page 84.
Fax Error Correction Mode. Error Correction Mode (ECM). The default is to leave
this feature enabled.
Hang up on Polarity Switch. Some countries, most notably Australia, signal remote
hangup by switching the polarity of the lines. This is sometimes called ROIC
(Reverse On Idle Condition). Only enable this option if you have confirmed with your
telco that hangups are signalled with polarity reversal.
Hardware Echo Cancellation. This option enables the hardware echo cancellation
module on BRI cards.
Hide caller id. Enabling this option will disable sending caller-id for this channel
group.
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Maximum transfer rate for fax transmissions. Maximum transfer rate used during
fax rate negotiation. The default maximum transfer rate is 14400.
Minimum transfer rate for fax transmissions. Minimum transfer rate used during
fax rate negotiation. The default minimum transfer rate is 2400.
Network Specific Facility Code. This option is rarely used. If you are unsure of what
to do, leave it set to None.
NSF, or Network Specific Facility codes are special codes that are sent in PRI facility
messages and are used to signal additional data to your telephone company. Do not
change this setting unless your telephone company explicitly stated that they require a
specific NSF value.
Number of Fax Buffers. Set the number of fax buffers. The default is 12.
This option lets you set a buffer policy to prevent fax failures on high latency systems.
See the related option “Fax Buffer Policy” page 43.
You must install a fax license and the fax software to make faxing available. For
details, see “Digium Addon Products” page 84.
Overlap dial. Overlap dialing sends each digit pressed as it's entered, instead of sending the entire number all at once. It is safe to leave this option disabled.
Pause Before Dialing. This option adds a slight delay to allow the off-hook signal to
propagate to the telephone company before sending down DTMF digits.
This should not need to be changed.
PRI Dialplan. This controls how outbound calls are sent over the PRI. Check with
your PRI provider to confirm which method they prefer.
PRI Reset Interval. The default value for this option, 3600, is usually correct.
This controls how often the PRI B channels are reset when they are unused. For the
vast majority of PRI lines, this has no effect on the PRI and is purely done just to be
safe. However, some PRIs in countries such as India may have problems with being
reset. In that case, you may want to enter a value of never.
PRI Switch Type. Your PRI provider should be able to provide you with the preferred PRI Switchtype. The most common option is National ISDN 2, sometimes
referred to as NI2.
Relax DTMF recognition. Enabling this option will make the DTMF interpreter
much more permissive. If you are having trouble receiving DTMF key presses in
IVRs with calls over a channel group, enabling this option may help. Unfortunately, it
also may result in many more false positives, or double keypresses. In cases where
poor line quality is hampering DTMF detection, it is highly recommended to improve
the line quality first before enabling this option.
Ring Debounce. This option is only used in very rare circumstances in certain countries.
This option controls how long Switchvox waits before confirming a ring on analog
lines. The value is in 1/8000ths of a second, and the default is 1024. Larger numbers
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cause Switchvox to wait longer, potentially making it harder to receive false-positives
or ghost calls.
Smaller numbers cause Switchvox to be more sensitive to rings.
RX Gain. We recommend enabling Automatic Gain Correction before altering this
option.
This controls the volume of audio received from this channel group. For example, if
incoming callers are too quiet or too loud, this option can change the volume. Values
are in decibels, and can be positive or negative to make calls louder or quieter respectively.
This is usually only needed on analog lines (not T1 lines) to compensate for signal
loss in the wires. Normal values range in the -15 to 15, and it's recommended to start
low, and increment the gains by a factor of 2 until they reach the correct volume (e.g.,
2, 4, 6, 8...).
Secondary Device Type. This option should almost always be set to PRI (Primary
Rate ISDN) CPE Side. Consult your T1 provider before selecting an alternate option.
Train echo canceller at beginning of calls. This option sends a pulse at the beginning of a call in the attempt to pre-train the echo canceller. This can improve situations with echo at the beginning of a call.
TX Gain. This controls the volume of audio sent from this channel group. It is recommended to not increase this value, although lowering it to -3 may improve call quality
and/or echo. If in doubt, leaving this setting at the default of 0.0 is recommended.
Use Caller ID. Enabling this option will enable receiving (and sending) caller-id.
It's almost always recommended to leave this set to Yes. The one situation where it
may be preferable to disable caller-id is when using analog lines in a country (such as
the USA) that sends the caller-id between the first and second ring. In order to receive
caller-id on analog lines, Switchvox has to wait until the second ring before answering. If caller-id is disabled, Switchvox will answer inbound calls immediately, without
waiting for the caller-id.
Use PRI t.309 timers.
No: 'pritimer=t309, -1'
Yes: 'pritimer=t309, 10000'
NOTE: Leave this option to the default of No.
Wait Duration. This forces the system to wait the specified number of seconds
before answering the call, allowing time to receive all of the call data before answering the call. The default is 1. Most likely you do not need to change the default.
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VOIP Providers
This tool lets you manage your Voice Over Internet Protocol (VOIP) Providers. VOIP
providers can be either commercial VOIP phone services or other VOIP hardware
such as a peered Switchvox. There are two common protocols used for placing phone
calls over the internet: SIP or IAX. If you are unsure which protocol to use, ask your
service provider.
From the VOIP Providers page, you can select either the Create SIP or Create IAX
Provider buttons and list and modify already created VOIP providers. You can also
download the current RSA Key (page 55) and set the RTP Port Range (page 56).
NOTE: To join two or more Switchvox PBXes as peered systems, use the Peered
Switchvox tool set. (See “Peered Switchvoxes” on page 56.)
SIP Providers
Click the Create SIP Provider button to display the create page with the Provider
Information tab opened. The other tabs are for advanced settings that you mostly
likely will not need to change from the defaults.
After you have created a provider, you are prompted to go to the Connection Status
page. This page displays your VOIP providers and their statuses. If Switchvox successfully contacts your provider, it is displayed as Registered. Now if you call the
number your provider gave you, your call should ring the Callback Extension. Or if
the status is Unreachable, you have the option of running diagnostic tests on the provider information.
SIP Provider Information Tab
SIP Provider Name. Create a name to refer to this provider. This is only used for
referring to this Provider in other areas of the Admin Suite.
Your Account ID. Enter the account ID provided to you by your SIP provider. This
may also be called Login, or Authentication ID.
Your Password. Enter your password for the account.
Hostname/IP Address. Enter the hostname or IP address for the SIP server that your
SIP provider gave to you. This may also be called Registrar Server.
Callback Extension. Enter the extension to ring when a voice call comes in over this
provider or channel. Callback extensions can be any extension type (your receptionist’s phone, an IVR, etc.).
Default Fax Extension. If you have fax service, enter the extension to ring when a
call identified as a fax comes in over this provider or channel. (Switchvox only identifies T.38 faxes over SIP providers. A fax coming in over an IAX provider or a channel, or a fax using a different standard may not be identified as a fax.)
NOTE: You must install a fax license and the fax software to make faxing available. For details, see “Digium Addon Products” page 84.
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DTMF Mode. DTMF tones are the sounds emitted when you press buttons on your
phone. Options in the dropdown include: Info, Inband, Auto, and RFC2833.
• SIP extensions, refer to your phone's user manual for the DTMF mode that your
phone uses.
• SIP providers, ask your provider which DTMF mode it supports.
• If you are unsure of which DTMF mode to select, use RFC2833 (the most common method).
Peer Settings Tab
NOTE: If you are peering two Switchvoxes together, do not create a VOIP Provider.
Instead create a Peered Switchvox, see “Peered Switchvoxes” page 56. The Peer Settings described here should no longer be used when setting up peered systems.
Peer Settings
Host Type. Provider is a host you must actively register to inform it of your location
and settings. Peer is another SIP PBX, including another Switchvox.
Treat system’s users like local users. If you want Switchvox to exchange Status
information with this peered Switchvox, set this option to YES.
Jabber Hostname. You can specify a hostname for the peered Switchvox’s Jabber
server. To find this hostname, look in the peered server's Network Settings page.
NOTE: This is important if you are peering Switchvoxes and want to use functions such as Presence or the Chat Panel.
Apply Incoming Call Rules to Provider. This option lets you alter the routing of
incoming calls.
• YES uses the default setting, which uses the rules on the Incoming Calls section as
well as the Callback Extension to decide which extension to ring when you receive
a call over this provider.
• NO allows callers from this peer to directly dial any extension in Switchvox.
Incoming calls will not follow the rules in the Incoming Calls section, and they
will not ring the Callback Extension.
Advanced Peer Settings. Click the arrow icon to this display the Host is a Switchvox PBX. If this VOIP Provider is a peered Switchvox, set to YES. This lets you use
advanced functionality with the peered Switchvox.
NOTE: This setting has been replaced by the Peered Switchvoxes feature. Go to
the Peered Switchvoxes section if you want to connect two Switchvox systems.
Passthrough Outgoing Call Rules. If you have selected NO for the Apply Incoming
Call Rules to Provider option, you can use this option to define how the remote peer
can place outgoing calls through this Switchvox.
Search for the Outgoing Call Rules, select the ones to add to the collection box
according the following options:
• To allow calls, put an Outgoing Call Rule into the collection as Allow.
• To deny calls, put an Outgoing Call Rule into the collection as Deny.
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• To quit and redisplay the list, select Cancel.
• To ignore an Outgoing Call Rule for this provider, do not put the Rule into the collection.
Caller ID Settings Tab
The Caller ID settings determine how your Caller ID is set on outgoing calls.
Supports Changing Caller ID. Changing your caller ID information may cause all
outbound calls to fail, so be careful when using this setting. Be sure that your provider
accepts a modified Caller ID, then enable this option and customize the Caller ID displayed on outbound calls.
Caller-ID method. Leave this set to From Header unless your provider explicitly
tells you to use a different setting. The vast majority of providers require caller ID to
be supplied in the From header.
Caller ID Name / Caller ID Number. Enter a name and phone number to display as
your Caller ID when making outbound calls with this provider.
WARNING: Setting this field may cause all outbound calls to fail; be sure that
your provider supports this feature. If your provider does not support this feature,
clear the Caller ID Name and Caller ID Number fields.
Connection Settings Tab
The Connection Settings determine how the connection is established with this provider.
SIP Port. This is the UDP port used to communicate with this SIP Provider. By
default this is set to 5060, the standard port for SIP communication. This port can be
changed if your provider or peeSr requires it.
SIP Expiry (in seconds). This controls the duration (in seconds) between SIP registration requests sent to your provider. Selecting a value other than 120 may cause
problems with NAT routers. Do not change this unless explicitly instructed by your
provider.
Proxy Host. If your SIP provider says you need to use a SIP proxy, enter the proxy
host IP here.
Authentication User. If your SIP provider says you need a different authentication
name than your account name, enter that authentication name here.
Always Trust this Provider. Most providers require this option be set to Yes. It is a
little less secure because it does not require authentication on incoming calls from this
provider. If you set this to No and it does not work properly, set it back to Yes.
NOTE: Leave this option set to Yes unless instructed otherwise. This option lets
the provider send you calls without authenticating. Most SIP providers do not
authenticate, therefore setting this to No may cause problems.
Qualify Hosts. Enabling this option sends a SIP Options packet to this SIP provider
every 60 seconds. This keeps NAT tunnels open. Response latency for these requests
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are shown in the System Status page. Note: Do not enable this option if your provider
does not respond to SIP Options packets.
Include user=phone in SIP. This option has no effect on most SIP providers. Enable
it if your provider explicitly requests it. This adds the string “user=phone” on the end
of the SIP URI used in various messages.
Use Local Address in From Header. Leave this set to No. Changing it to Yes will
violate section 8.1.1.3 of the SIP standard, RFC 3261. Only set this to Yes if your provider explicitly requests that you violate the SIP standard in this manner. If this is Yes,
Switchvox uses its own local IP address in the From header of requests.
SIP Provider Host List
Some SIP providers have multiple servers on the internet that communicate with
Switchvox. For example, you may register sip.example.com but receive calls from
ny.example.com, la.example.com, and mia.example.com. Use this option to list multiple hosts (or IP addresses) so that Switchvox can route calls from them. If you do not
know if there are multiple hosts, ask your provider.
If there are multiple hosts, ask your provider to supply you with a list of their IP
addresses.
Call Settings Tab
The Call Settings determine how to manage calls with this provider.
Audio Codecs. Some providers support more than the normal audio codecs. These
additional codecs may provide compression or other features that save bandwidth or
lead to better call quality. However, you should check with your provider before
selecting another codec, to be sure it is supported. If your Switchvox is a peer with
another Switchvox, you can choose any audio codec.
The following table describes the audio codecs. If you entered a license key for the
G729 Codec in Tools > Digium Addon Products, you will also see G729 in the list
of codecs.
Codec
Bandwidth
Usage (kbits/s)
Sound Quality
CPU Usage
ULAW
85
Great
None
ALAW
85
Great
None
G722
85
Excellent
Low
G726
54
Good
Low
G729
30
Good
Medium
SPEEX
33
Great
High
GSM
35
Good
Low
ADPCM
54
Good
Low
LPC10
22
Poor
Medium
Video Codecs. The video codecs H.263, H.263+, and H.264 are available.
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NOTE: Video calling is not available over IAX, so these codec settings are not
available in the IAX VOIP Provider section.
Map Distinctive Rings. Some providers let users set different ring tones based on the
dialed number, but do not pass the dialed number information directly. You may have
to experiment with this setting and the settings of your SIP provider to get the right
phone number associated with the right ring type.
Call Settings.
• Enable Jitterbuffer. Enable this option to combat network problems. A jitterbuffer induces a small delay in call latency to make up for lost or delayed voice packets. You may want to enable the jitterbuffer if you have bad network connectivity
between Switchvox and the Provider.
The options are as follows:
-Never: It’s best to leave the jitterbuffer disabled if you are not having any audio
quality problems.
-When Needed: Intelligently enable the jitterbuffer with a dynamic window size
when the call is terminated on Switchvox. Try this option first if you have audio
quality issues.
-Always: Always enable the jitterbuffer with a dynamic window size. This may
be useful if you are using phones or other SIP devices that do not have their
own jitterbuffer.
-When Needed with Fixed Window: Intelligently enable the jitterbuffer with a
fixed window size. This may be useful if jitter is sporadic, yet sudden. Or, if
you need to mask problems with poorly configured routers that cannot implement QoS.
-Always with Fixed Window: Always enable the jitterbuffer with a fixed window size. Select this if all else fails. Note that this just masks over the underlying issue. It’s best to fix the jitter problem itself, instead of using this option.
• Allow Reinvite. We strongly recommend leaving this set to Never. Enabling this
option is risky and will likely result in sporadic one-way audio, dropped calls, or
not being able to place calls. This lets Switchvox redirect the media stream of
some calls, bypassing the PBX and passing directly between both participants.
This may offer bandwidth and quality improvements in certain scenarios, but it is
extremely risky.
-Never: This is the safest setting that presents the least likelihood of trouble.
-Always: The PBX directs media streams between participants when it is able
to. Set this at your own risk.
-Non-NAT Connections Only: The PBX redirects media streams only between
participants who are not behind NAT.
-Always, Using the UPDATE method: This is the same as the Always setting,
but uses a different mechanism within the SIP protocol. Only set this if you
have been explicitly instructed to do so.
-Non-NAT Connections Only, Using the UPDATE method: The PBX redirects media streams only between participants who are not behind NAT, using a
different mechanism within the SIP protocol.
IMPORTANT: This setting is just a recommendation to Switchvox that reinvite
is possible. There are many scenarios, such as when a call is being monitored or
recorded, where Switchvox will not be able to reinvite the call away.
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• Always Send Early Media. Setting this option to Yes causes Switchvox to send
RTP audio before a call is answered. Enabling this option may resolve an issue
where audio is not sent by certain SIP providers when an inbound call is cascaded
out of the same provider.
• Voicepulse Connect DID Workaround. This option is a workaround for SIP providers that send two INVITE messages for every phone call to handle SIP devices
that handle DIDs and ones that do not. Only enable this option if you are sure your
provider is sending two INVITEs for every call: one with the DID in the From
header, and one with your account ID in the From header.
Fax Settings Tab
The Fax Settings determine how faxes are handled with this provider. You must install
a fax license and the fax software to make faxing available. For details, see “Digium
Addon Products” page 84.
IMPORTANT: We recommend that you only send T.38 faxes over SIP. We do
not recommend sending faxes over IAX connections. Switchvox cannot necessarily recognize a fax over an IAX connection.
• Error Correction Mode for G711 Faxes. The default is to leave this feature
ENABLED.
• Minimum transfer rate for fax transmissions. Minimum transfer rate used during fax rate negotiation. The default minimum transfer rate is 2400.
• Maximum transfer rate for fax transmissions. Maximum transfer rate used during fax rate negotiation. The default maximum transfer rate is 14400.
• Maximum expected T38 packet delay. The result of this specification is that all
T38 packets are padded to prepare the fax stack for the specified delay. As long as
the packet delay is less than or equal to the specified value, the fax stack will not
realize that there is an actual packet delay because of the packet padding. The
default value of 800 milliseconds should cover most use cases.
Fax Detection. This option describes the method of fax detection.
• YES. Detects incoming faxes based on the receipt of a CNG tone or a T.38 invite.
• CNG. Detects incoming faxes based on the receipt of a CNG tone.
• T.38. Detects incoming faxes based on the receipt of a T.38 invite, without CNG
tone.
• NO. Does not detect incoming faxes.
NOTE: You must install a fax license and the fax software to make faxing available. For details, see “Digium Addon Products” page 84.
Error Correction Mode for T.38 Faxes. Detects incoming faxes based on the
You must install a fax license and the fax software to make faxing available. For
details, see “Digium Addon Products” page 84.
Error Correction Mode for G711 Faxes
The default is to leave this feature ENABLED.
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Initiate T38 reinvite after timeout
This option initiates a T.38 re-invite if it is not received within 10 seconds. The
default is YES.
Enable Audio Failover
This option causes Switchvox to revert to audio fax mode in the event that T.38 negotiation fails. The default is YES.
IAX Providers
Click the Create IAX Provider button to display the create page with the IAX Provider Information tab opened. The other tabs are for advanced settings that you mostly
likely do not need to change from the defaults.
After you have created a provider, you are prompted to go to the Connection Status
page. This page displays your VOIP providers and their status. If Switchvox successfully contacts your provider, it is displayed as Registered. Now if you call the number
your provider gave you, your call should ring the Callback Extension. Or if the status
is Unreachable, you have the option of running diagnostic tests on the provider information.
IAX Provider Information
IAX Provider Name. Any string to identify this VOIP provider. This is only used for
referring to this VOIP Provider in other areas of the Admin Suite.
IAX Provider Label. Label this provider gives itself when placing calls to your system. This is usually the name inside of [ ] brackets in Asterisk configuration examples.
Your Account ID. Username provided by IAX Provider to authenticate your Switchvox PBX to the provider during IAX registration and outbound calls.
Your Password. Your password for the account.
Hostname/IP Address. Primary hostname or IP address of your provider's IAX
server. This hostname will be used to send registration requests. It will also be the host
that will send any inbound calls to your PBX.
If your provider has multiple servers for their IAX service, you can use the Primary,
Secondary, and Tertiary Outbound Host settings to list them. Otherwise, the Hostname/IP Address field will be used for placing outbound calls as well as inbound calls
and registration requests.
Callback Extension. The extension to ring when a voice call comes in over this provider or channel. Callback extensions can be any extension type (your receptionist’s
phone, an IVR, etc.).
Incoming Authentication
Incoming Authentication Type. Select one of these choices from the dropdown:
MD5, RSA, or plaintext. MD5 and plain text use your password for authentication. It
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is preferable to use the MD5 method over the plain text method. RSA uses public and
private encryption keys to authenticate the provider. If you select this option, enter the
public RSA key in the RSA Authentication Key field.
Use Existing RSA Key. Select from the dropdown of available keys,
or Upload New RSA Key. Upload a new key from your computer hard drive.
Peer Settings Tab
The Peer Settings determine the relationship with this provider.
Host Type. This setting controls registration between this Switchvox and the remote
host.
• Provider is the default setting where this Switchvox sends registration to the
remote host.
• Client is the setting where the remote host is the client and it sends registration
requests to this Switchvox.
• Peer is the setting for peered hosts that do not need to register. If you choose Peer,
make sure that both hosts have static IP addresses.
Treat system’s users like local users. If you want Switchvox to exchange presence
and line-use information with this peered Switchvox, set this option to YES.
Jabber Hostname. You can specify a hostname for the peered Switchvox’s Jabber
server. To find this hostname, look in the peered server's Network Settings page.
This is important if you are peering Switchvoxes and want to use functions such as
Presence or the Chat Panel.
Apply Incoming Call Rules to Provider. This option lets you alter the routing of
incoming calls.
Yes is the default setting. It uses the rules on the Incoming Calls section as well as the
Callback Extension to decide which extension to ring when you receive a call over
this provider.
Select No if this provider is a peer. This allows callers from this peer to directly dial
any extension in Switchvox. Incoming calls will not follow the rules in the Incoming
Calls section and they will not ring the Callback Extension.
Advanced Peer Settings. Click the arrow icon to this display the Host is a Switchvox PBX. If this VOIP Provider is a peered Switchvox, set to YES. This lets you use
advanced functionality with the peered Switchvox.
NOTE: This setting has been replaced by the Peered Switchvoxes feature. Go to
the Peered Switchvoxes section if you want to connect two Switchvox systems.
Passthrough Outgoing Call Rules. If the Apply Incoming Call Rules to Provider
option is No, use this section to define how the remote peer can place outbound calls
through this PBX. Use the check boxes to assign which outgoing rules apply to this
peer. Check the Allow box to apply the rule to this peer's inbound calls. Check Deny
to block all calls that match that rule, such as blocking all long-distance numbers. To
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bypass a rule, leave both options unchecked. See “Outgoing Calls” page 58 for more
information about each outgoing rule.
Caller ID Settings Tab
Caller ID Name / Caller ID Number. Enter a name and phone number to display as
your Caller ID when making outbound calls with this provider.
WARNING: Setting this field may cause all outbound calls to fail; be sure that
your provider supports this feature. If your provider does not support this feature,
clear the Caller ID Name and Caller ID Number fields.
Connection Settings Tab
The Connection settings determine how the connection is established with this provider.
Primary, Secondary, Tertiary Outbound Hosts. Some IAX providers use different
hosts for registration, incoming, and outgoing calls. If your provider has multiple
hosts for placing outbound calls, you can use the Primary, Secondary, and Tertiary
Outbound Host fields to specify them. If these are left blank, the PBX will use the
host specified in the Hostname/IP Address field for outgoing calls.
Incoming Password. If this provider uses passwords for incoming call authentication
and the password is different from your account password, specify it here. This condition is rare unless you are peering with a remote host and have explicitly configured
the inbound and outbound passwords to be different.
Qualify Hosts. Enabling this option will periodically send IAX POKE packets to the
remote host and time their response. This latency measurement can then be seen in the
System Status page.
Enable Trunking. Trunking can be used to reduce the bandwidth used across multiple simultaneous calls; however, it must be enabled on both sides of the link. This is
useful when you have two systems peered together over IAX with heavy voice traffic.
If using commercial IAX services, check with the provider to see if it supports trunking. Enabling trunking on only one host will result in one-way audio on phone calls.
Use RSA Authentication for Outgoing. Before setting this option, place your public
RSA key on the remote host. You can retrieve this system's public RSA key in System
Setup> VOIP Providers.
Call Settings Tab
The Call settings determine how to manage calls with this provider.
Audio Codecs. Some providers support more than the normal audio codecs. These
additional codecs may provide compression or other features that save bandwidth or
lead to better call quality. However, you should check with your provider before
selecting another codec, to be sure it is supported. If your Switchvox is a peer with
another Switchvox, you can choose any audio codec.
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The following table describes the audio codecs. If you entered a license key for the
G729 Codec in Tools > Digium Addon Products, you will also see G729 in the list
of codecs..
Codec
Bandwidth
Usage
(kbits/s)
Sound
Quality
CPU Usage
ULAW
85
Great
None
ALAW
85
Great
None
G722
85
Excellent
Low
G726
54
Good
Low
G729
30
Good
Medium
SPEEX
33
Great
High
GSM
35
Good
Low
ADPCM
54
Good
Low
LPC10
22
Poor
Medium
Enable Jitterbuffer. Enable this option to combat network problems. A jitterbuffer
induces a small delay in call latency to make up for lost or delayed voice packets. You
may want to enable the jitterbuffer if you have bad network connectivity between
Switchvox and the Provider.
Fax Settings Tab
The Fax settings determine how faxes are handled with this provider.
You must install a fax license and the fax software to make faxing available. For
details, see “Digium Addon Products” page 84.
IMPORTANT: We recommend that you only send T.38 faxes over SIP. We do
not recommend sending faxes over IAX connections. Switchvox cannot necessarily recognize a fax over an IAX connection.
Error Correction Mode for G711 Faxes. The default is to leave this feature
enabled.
Minimum transfer rate for fax transmissions. Minimum transfer rate used during
fax rate negotiation. The default minimum transfer rate is 2400.
Maximum transfer rate for fax transmissions. Maximum transfer rate used during
fax rate negotiation. The default maximum transfer rate is 14400.
RSA Key
RSA keys can be used for authenticating peers using IAX. This is not usually
necessary when using commercial IAX providers, but it can be handy when peering
multiple IAX-compatible PBXs together. Click Download to save the current RSA
key. Then you can upload it to other phone systems when you create or modify an
IAX Provider on that system.
You can rename this RSA key to something unique to this Switchvox PBX: Enter a
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new name in the Rename RSA key field, then click Rename. This can help you
identify keys as you set up peered PBX systems. For example, you may name this key
bostonoffice and another system’s key londonoffice.
RTP Port Range
These are the source ports used for establishing the RTP media pathway for SIP calls.
It is unlikely that you will need to change this setting. Only change it at your
provider’s request.
Peered Switchvoxes
This tool allows you to configure a Switchvox Peer and to manage extension groups
that you may want to share with a Switchvox Peer. This includes viewing extensions
that are duplicated on a peer, setting a dialing prefix if there are duplicate extensions,
adding a peer's extensions to the Internal Directory, and checking the system status of
peered Switchvoxes.
To peer two Switchvox PBXes, each PBX must be externally accessible to the other,
and each must have its own unique hostname. Username and password are the same
for peered systems.
Configure New Peer
From the Peered Switchvoxes page, click Configure New Peer to display the Connection and Routing tab. Click Save Peered Switchvox when finished configuring
or modifying.
Connection and Routing Tab
Display Settings
• Peer Name. Enter a unique name for the first Switchvox.
Connection Settings
• Peer Host Address: Enter a unique hostname or IP address for the first Switchvox.
• Shared Username. Enter the username, which will be the same for each Switchvox in this peered group.
• Shared Password.Enter the password, which will be the same for each Switchvox
in this peered group.
• Peer’s SSL Certificate in PEM Format. Click the Fetch Certificate icon to
download a certificate.
Advanced Connection Settings (arrow toggle)
These settings typically do not need to be changed.
• SIP Port. This option is preset to 5060.
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• HTTPS Port. This option is present to 443.
• Alternate Incoming Host Addresses (Hostname or IP). Enter an alternate Hostname or IP.
Call Settings
• Dial Prefix. Enter any prefix needed. This is used to create the outgoing call rule
Advanced Call Settings (arrow toggle)
NOTE: These settings typically do not need to be changed.
• Call Quality: Enable Jifferbuffer.
• Passthrough Outgoing Call Rules.
Audio Codecs. See table under SIP Providers > Call Settings.
Video Codecs. See table under SIP Providers > Call Settings.
Published to Peer tab
Publishing an extension group means the item is displayed in the user interface of the
peer. If someone on the peer does some form of search, published items can be
returned.
Extension Groups
Publish All Groups. All current and future Extension Groups. Select YES/NO.
If YES, you cannot add any extension groups to the collector because all groups are
published.
If NO, you can use Type to Search to find extension groups. Select the extension
groups you wish to publish and click the Add button.
NOTE: Unpublished extension groups can be reached from the peer, they are just
not returned in a search.
View Currently
Configured Peers
If configured peers are available, clicking the Peered Switchvoxes option displays
them in a table with following information:
•
•
•
•
•
Name. Name of the peer.
Hostname. Peer hostname or IP address.
Dial Prefix. Peer’s dial prefix setting, if any.
Shared Username. Shared username.
Actions. Icons for the following:
-Content published. If the reciprocal peer connection is successful, clicking the
this icon displays Content Published from the Peer, including Extension Groups
and Extensions.
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-Modify. Clicking this icon displays the settings already defined for the peer that
can now be modified..
-Delete. Clicking this icon displays a message that informs you that clicking
the Yes, Delete button will permanently remove this peered Switchvox and its
associated data. Clicking the Cancel button removes the deletion message and
buttons.
Local Peer Settings are also displayed.
Local Peer Settings
Name. Local Server is the default.
Clicking Save Local Settings saves.
Outgoing Calls
The Outgoing Calls tools let you handle all aspects of outgoing calls. It is comprised
of three tabs: Outgoing Call Rules; Caller ID Rules; and Call Diagnostics. All rules
are displayed in a table called Outgoing Call Rules. Clicking the Create Outgoing
Call Rule button displays the Create Outgoing Call Rule page.
Outgoing Call Rules
An outgoing call rule is a dialing pattern and associated routing information. For
example, the International outgoing call rule is triggered when a user dials “9011” followed by 7 to 13 digits. Such calls are sent to the selected route (such as a VOIP provider or a Channel Group).
You can create multiple call rules that match the same pattern. In this case, if there is a
pattern match, the rule that is highest in the priority list gets called first. If that rule
fails, and it is not marked as a final rule, the next rule is called, and so on.
Switchvox is preconfigured with a set of outgoing call rules that serve most situations.
You can modify or delete those Rules, and create your own.
Rule Name & Note
Rule Name. A unique name for this rule.
Rule Note. A descriptive note for this item. This is displayed when you hover over
the note icon.
Pattern to match
Enter the call rule information in the boxes as follows:
• Number begins with the digits [ ]. Enter any required precall numbers; for
example, Switchvox normally requires dialing a 9 for an outside call. In this case,
enter 9 in this box. After the precall digit, enter the pattern you want to match. You
can use any digit 0 through 9, pound (#), asterisk (*), and the plus sign (+) to
describe a pattern. If a call uses the pattern exactly as described, then the rule is
applied. You can also match multiple patterns in the same rule by using parenthe-
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sis and the pipe (|). To do this, put all of the possible patterns between parentheses
and separate them with a pipe. The default rule Toll Free is a good example:
91(800|888|877|866). If a call begins with 91800, 91888, 91877, or 91866, then
the rule is applied.
• The rest of the number must be between [ ] and [ ] digits in length. Enter the
length of a telephone number; for example, a local number must be exactly 7 digits. In this case, enter 7 in both boxes.
• Before connecting the call, trim [ ] digits from the front. Enter the number of
digits to trim when dialing; for example, enter 1 to remove the prepended 9 from
the number sent to the provider.
• and then prepend the digits [ ] to the number. Enter the digits to prepend to the
telephone number; for example, enter 1 and the area code for the number.
Call Through
This indicates which of your Channel Groups or Providers to use for this call, or it's
an internal call (within Switchvox).
Primary Call Through Provider. Select from the dropdown menu the provider or
group that is the first call through.provider. Clicking the Info icon displays the attributes of the selection.
Failover Call Through Providers. Click the Add button to add up to 5 more providers if Switchvox cannot place the call through the previous provider. This helps you
protect against downtime if one of your providers is experiencing technical difficulties. Switchvox tries to complete the call in the order that the Failover Providers are
shown.
Rule Final
Is this rule final?
• If YES, when a call matches this rule but the route fails, then Switchvox does not
look for additional matching rules and the call is not completed.
• If NO, when a call matches this rule but fails, Switchvox looks for another matching rule, and the call might be completed.
NOTE: One reason the routing might fail is if the extension is denied permission
to use the outgoing rule.
Caller ID Rules Tab
Caller ID Rules let you modify the outgoing caller ID name and number.
Caller ID rules can apply to:
• A single extension.
• Any extension. (Use a Single Rule, and put the word any in the Select Extension
field.)
• A range of extensions.
• An extension group.
If you use a ranged Caller ID Rule, you can derive an extension’s correct caller ID by
adding and prepending numbers to the extension.
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Call Routing
For example, if you have an extension block from 100 to 200, and a DID block of
800-555-1200 - 800-555-1300, you would use the following ranged rule:
When an extension is in the range from 100 to 200 and is making an outgoing call
through rule Any Outgoing Rule, change their caller id to a number derived by adding
100 to their extension and prepending 8005551.
NOTE: Not all VOIP providers support this feature and some VOIP providers do
not work properly with altered Caller IDs. Also, it is not possible to set Caller ID
on regular analog phone lines.
Call Diagnostics Tab
Outgoing Call Diagnostics can be used to determine which Outgoing Call Rule is
being used and why.
Enable Diagnostics. YES reads diagnostic information over the phone to all callers
from all extensions. For example, If you were to Enable Diagnostics and then dial an
internal extension (800), instead of connecting the call, the system would read back,
“You have dialed, eight zero zero. This call will be sent to outgoing rule number
seven. This rule will send the call to extension eight zero zero. Press one to attempt
this call through rule number seven.” As outgoing calls become more complex by
trimming or appending digits, altering Caller ID, or matching multiple outgoing call
rules, the diagnostic information provided will become more complex.
Listening to the diagnostic information and following the voice prompts provides
insight into the path that calls are taking in the system to solve calling problems or
unexpected call behavior taking place in the system. For example, the diagnostic may
reveal that a call is using the wrong rule.
When using the Diagnostic mode, make sure the number you are dialing matches the
pattern for the rule. If the pattern does not match, your outgoing call will not be connected using that rule. If the number you have dialed does match the pattern for the
rule, make sure the extension you are placing the call from has permission to call
through that rule (see “Outgoing Call Rules” page 58).
Incoming Calls
The Incoming Calls tools let you manage incoming call rules and routes via two tabs:
DID Routes, the default; and Caller ID Rules. A DID Route maps numbers coming in
on a particular Provider or Channel Group to a particular extension., and Incoming
Call ID Rules let you perform actions on calls based on who is calling you. These
rules are system wide and happen before any of the extensions' call rules are triggered.
DID Routes Tab
By default, Switchvox creates a DID route (the callback extension) for each interface
as they are created (i.e., each provider has a rule). You can add new rules that allow
for specialized routing based on information sent by the provider or deduced from the
channel group.
To create a new DID route, click Create Single DID Route or Create Ranged DID
Route. A Single DID Route always routes calls to a single, specific extension. A
Ranged DID Route derives the correct extension number from the DID.
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Call Routing
You can put all of the DID Routes in order, to ensure that call reach the correct extension.
NOTE: Some SIP providers and most types of analog lines do not send the number dialed to the PBX. Ask your phone service provider if they support multiple
incoming numbers or DIDs.
Create Single DID Route
This option lets you send a single phone number to a specific extension. For example:
Route number 8005551212 on numbers from Channel Group T1 to extension 802
Single DID Route Settings. enter/specify the following information:
•
•
•
•
•
•
•
Rule Name to identify the rule.
Note. Clarifying information regarding the rule.
Incoming DID to Match.
Incoming Provider. Select provider from the list of existing Providers.
Incoming Call Type. Select from dropdown options: Voice Calls, Faxes, or both.
Extension to Route Call. Search for or enter the extension to route.
Process All Calls as a Fax. If you have fax capabilities, and you have selected
both types of calls, you call indicate YES; otherwise, this option is not available.
NOTE: You must install a fax license and the fax software to make faxing available.
For details, see “Digium Addon Products” page 84.
When finished, click Save Single DID Route.
Ranged DID Route
This option lets you send a range of phone numbers to an extension that is derived
from the phone number itself. This is useful if each person has an outside number for
directly dialing their extension. For example: All Voice Calls on numbers from Channel Group T1 ranging from 8005551200 through 8005551300 will route to an extension derived by trimming 7 digits from the front and adding 0 to the result.
Ranged DID Routhe Settings. Enter the following information:
•
•
•
•
•
Rule Name to identify the rule.
Note. Clarifying information regarding the rule.
Beginning Range of Incoming Call. First number in the range of numbers.
End Range of Incoming Call. Last number in the range of numbers.
Incoming Call Type.Type of calls to affect: Voice Calls, Faxes, or Voice Calls and
Faxes.
• Incoming Provider. Select provider from the list of existing Providers.
• Number of digits to trim. Enter the number of digits to trim from the front of the
phone number.
• Derive Resulting Extension by adding this to the trimmed call. Enter a value to
add to the result or you can leave this blank.
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Caller ID Rules Tab
To create a new Incoming Caller ID Rule, click either Create Call Transfer Rule or
Create Block Number Rule.
Create Transfer Call Rule
This option lets you create a rule that transfers certain callers to a specific extension.
For example, you could route all calls from a vendor whose phone number is
8005554321 to the extension for your purchasing department.
NOTE: All phone numbers used in an incoming call rule must be exactly as they
appear in your caller ID. This number is usually the full 10-digit number, which
includes the area code.
To create a new incoming call rule, enter/select the following information:
•
•
•
•
•
•
Rule Name to identify the rule.
Note. Clarifying information regarding the rule.
Incoming Number Type from the dropdown. Options are Phone Numer or Prefix.
Incoming Number. Enter the phone number or prefix that you want to affect.
Extension to Transfer to. Enter the extension the call should be transferred to.
Timeframe. Select Any Time, During Business Hours, Outside Business Hours,
or Weekends Only from the dropdown. (For more information, see “Time
Frames” page 80.)
When finished, click Save Call Transfer Rule and a message is displayed indicating
the new rule was successfully saved.
Create Block Number Rule
This option lets you block callers from certain phone numbers or prefixes. For example, a rogue fax machine keeps calling your office with a caller ID number of
8005551234. You could add a Block Number incoming call rule that hangs up anytime this number calls you. You can block whole prefixes of numbers by choosing to
block Prefix rather than Phone Number.
Enter/select the following:
•
•
•
•
•
Rule Name to identify the rule.
Note. Clarifying information regarding the rule.
Incoming Number Type from the dropdown. Options are Phone Numer or Prefix.
Incoming Number. Enter the phone number or prefix that you want to affect.
Block Action. Enter the blocking actions from the dropdown options: Play Busy
Signal, Play Congestions, or Hang Up.
• Timeframe. Select Any Time, During Business Hours, Outside Business Hours,
or Weekends Only from the dropdown. (For more information, see “Time
Frames” page 80.)
When finished, click Save Block Number Rule and a message is displayed indicating
the new rule was successfully saved.
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Modifying Caller ID Rules
To change an incoming call rule, modify the settings for the rule and click its Save
button. The rule is saved and a message is displayed indicating the rule was successfully saved.
Deleting Caller ID Rules
To delete an incoming call rule, click its Delete button. The rule is deleted permanently and a message is displayed indicating the rule has been deleted
Admins
The Admins tools give you a means to manage Switchvox accounts:
Manage
63
General Information Tab 63
Permissions Tab 64
My Account
64
Language and Locale 64
New Password 64
Retype New Password 64
Minimalist Mode 64
Manage
You can create multiple 'sub-admin' users, each with their own permissions. This way,
you can assign Switchvox administrative tasks to various people and limit their access
to just the tasks they need to perform.
Click Create Admin to display the Create Admin page.
General Information
Tab
Enter a new Login name for the admin, then enter and re-enter a new password. The
password is case-sensitive, and it can contain special characters. The strength of the
new password is indicated, to help you create a good password.
Leave the box checked for Force Password Change at next login, unless you really
want your new sub-admin to use the password that you created. If the sub-admin creates a weak password, Switchvox shows you with an alert.
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Permissions Tab
You can set this sub-admin’s access for each of the main sections in Switchvox. You
can block the sub-admin from using a section at all, or let him or her see the information in that section, or make changes (which allows full access for that section).
Click Save Admin, and the Manage Admins page is displayed with the new subadmin included in the Admins list.
To modify a sub-admin, click its Modify icon.
To delete a sub-admin, click its Delete icon.
My Account
This is the information for your administration account.
Language and Locale
The Switchvox web-based tool suite is available in several languages:
English for United States
English for United Kingdom
Spanish for Spain
Spanish for Mexico
Italian
If you choose to use a language other than English for US, be aware that all other languages and locales handle dates with the format of dd/mm/yyyy (as opposed to mm/
dd/yyyy).
New Password
The password for this account. Leave blank if there are no changes.
Administration accounts can have extensive control over Switchvox, so you should
use a very strong password to protect this account. Strong passwords are fairly long,
combining letters, numbers, and special characters.
Retype New Password
For verification, enter the password again.
Minimalist Mode
This mode presents the Web Suite interface with reduced graphics.
If Minimalist Mode is set to YES:
• The background graphic and larger graphics on buttons and tabs are removed.
• Some text, especially in section headers, is smaller.
• Animations are removed.
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Tools
This section describes the functions under the Tools category, which are used to set up
and manage the PBX tools such as Time Frames, IVR Menus, Voicemail, and Music
on Hold.
PBX Features 62
Notifications 89
Extension Features 91
Media 96
PBX Features
Interactive Voice Response (IVR) Editor 63
Managing IVR Menus 63
IVR Actions 66
Time Frames 78
Call Recording 79
Digium Addon Products 82
Event Triggers 82
Voicemail 85
SMTP Settings 87
Notification Templates 87
Contacts 90
Distinctive Ringtones 91
Switchboard Panels 91
Digium Phone Apps 94
Music On Hold 95
Sound Manager 97
Interactive Voice Response (IVR) Editor
IVR is an automated menu system (also known as an auto-attendant) that you can
program to greet and interact with your callers. An IVR can provide information to
the caller, collect information from the caller, and send the caller to the right
extension. IVR Editor is the entry to this process.
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Managing IVR Menus
Select the IVR Editor from the PBX Features section. The IVR Editor page is comprised of a Create IVR Menu button and the following sections for any default or
already created IVR menus: IVR Tree, IVR Menu Details, Actions, and Options,
and a dropdown menu to select an IVR to view or modify.
See the following figure..
Planning a New IVR
Experience
Before creating an IVR Menu, you should plan in advance the entire IVR experience
you want to offer your callers. Then create the sequence starting from the end and
working towards the beginning as is shown in the following example:
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• Determine all of the recorded sounds you want to play for the caller (instructions,
information, etc.) and create those sounds in the Sound Manager (see page 98).
• Determine which extensions the caller might end up being transferred to and create an IVR menu for each one. (Those menus would have a Dial Extension action,
see page 73.)
• Determine the factors that might change your IVR and build an IVR menu to handle each situation. For example, if your IVR behaves differently during regular
business hours than after hours, or if it behaves differently based on a zip code the
caller provides, you need to create an IVR menu for each situation.
After you have your sounds and a set of component IVRs built, then create the Main
IVR Menu that answers your calls.
Create IVR Menu
Click Create IVR Menu to display its menu page.
• Enter a unique name and a description for the new IVR.
• Click Save IVR Menu.
This displays the IVR Editor page with a message indicating the new IVR Menu
was created. The new menu is displayed in the IVR Tree and IVR Menu Details.
After you are done building your main IVR, you need to create an extension that
reaches it. For details, see IVR Extensions. Also, if you want all calls from your
phone service provider to go to your main IVR, you can set that provider’s Callback
Extension to the extension of your main IVR.
Modify IVR Menu
To modify an existing IVR Menu, select it from the IVR Menu to Modify: IVR
Menu Name (Extension) dropdown menu. This menu presents your IVRs in two
parts: IVRs that have extension numbers and IVRs that do not have extension
numbers. After selecting an IVR Menu, click the View button to update the IVR Tree
and IVR Menu Details, including Actions and Options, and the Description for the
selected menu.
IVR Tree
IVR Tree displays an overview of the selected menu. IVR Actions are listed first
with various icons to indicate the kind of action. The last action in the list is always
Listen for options, which cannot be modified or deleted. This is the action that
handles the IVR Options.
IVR Options are exposed by expanding their respective folder icon. Options are an
entry point into another IVR.
Click on any action, folder, or option name to see details in the IVR Menu Details.
IVR Menu Details
The first section displays the IVR Menu Name, its Description, and the Modify IVR
Menu and Delete IVR Menu buttons.
Actions. Lists each action in the IVR in detail, followed by a Modify Actions and
Create Action buttons.
Options. Lists each option in the IVR in detail, followed by a Modify Options and
Create Option buttons.
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IVR Actions
You can add, modify, or delete actions associated with an IVR Menu. Most actions
have a configuration page on which you can specify the details of the action, then save
your changes.
Adding an Action to
an IVR Menu
Modifying an Existing
Action
Deleting an existing
action
Actions Types
1.
Click either Create Action or Modify Actions from the IVR Menu Details. This
displays either the Create IVR Action page or the Modify IVR Actions page.
2.
If modifying, click the Create IVR Action button to display the Create IVR
Action page.
3.
Select an action from the Action Type dropdown menu. After you make a selection, the Action Description displays a short description of the selected action.
4.
Click Next.
5.
Complete the detail of the action as appropriate, then click Save Settings.
1.
Click Modify Actions in the IVR Menu Details. This displays the Modify IVR
Actions page.
2.
Click the Modify icon for the action you want to modify. This displays the Modify Action page for that action.
3.
Make your changes and click Save Settings.
1.
Click Modify Actions in the IVR Menu Details. This displays the Modify IVR
Actions page.
2.
Click the Delete icon for the action that you want to remove. This displays the
Delete Action page for that action. Click Yes, Delete Action.
Interruptible Actions
Some IVR actions can be interrupted if the caller presses a key on his or her keypad.
Although this can offer a better caller experience, you must plan for it carefully
because it has repercussions throughout your IVRs. Take care about of order of
actions in your IVR and where interruptions are allowed.
Here are the rules Switchvox follows for interruptions in IVRs:
1.
If an action is not interruptible, all keys pressed before or during that action are
dropped. If the caller pressed keys during previous interruptible actions, those
keys are lost.
2.
If an action is interruptible, the caller can press any key to interrupt, and the IVR
remembers that one key.
3.
If multiple subsequent actions are interruptible, the IVR remembers each key that
was pressed to interrupt each action.
4.
When an IVR has collected one or more keys through a series of interruptible
actions, it uses those keys (in the order they were pressed), in the following manner:
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1.If
there is an upcoming Record Digits action, with no un-interruptible action in
the way, the keys are reserved for the Record
2.Digits
action. They will not be used to dial an extension or to select an option.
3.If
Enable Extension Dialing is on, and the first key pressed matches an extension number, Switchvox pauses for 3 seconds to see if the caller will press a
second key. If a second key is pressed and a valid extension is dialed, the call is
transferred. However, if two or more keys are pressed that do not match a valid
extension, then the key press is considered an Invalid Option (see IVR
Options).
4.If
Enable Extension Dialing is not on, or the caller presses a key that does not
match an extension, or the caller does not press a second key within 3 seconds,
the option that matches that key is selected and the caller is sent to that option’s
IVR menu.
Play Sound
This action plays a sound from the Sound Manager to the caller. If you want to play
sounds recorded by the caller, use the action ‘Play Recorded Sound.’
• Folder. Select the Sound Manager folder where this sound is located.
• Language. Select the language for this sound. Current Language is the default and
is the language that was previously set in the IVR. The default is English.
• Sound to Play. Select the sound to play.
• This sound is interruptible. Check this box if this is interruptible. For details, see
“Interruptible Actions” ( page 68).
• Play Sound: Advanced Options. If you want to decide which sound to play based
on a sound ID stored in a variable, click on the Show Advanced Options link. This
lets you select variables that have been previously created using the ‘Set Variable’
action. Make sure that the value in the variable is a sound ID from the Sound Manager, or this action will not be able to play the sound.
Record Sound
This action records a sound from the caller, and saves it in a variable. The sound
variable can be used later on in the IVR menu for actions such as playing or
uploading.
When a caller reaches a Record Sound action, Switchvox plays a beep and then begins
recording. Recording stops when the caller presses the "#" key, hangs up, has
recorded too much silence, or has reached the total minutes of recording time.
• Stop Recording after sections of silence. Enter the number of seconds of silence
allowed before recording stops.
• Stop Recording after total minutes. Enter the maximum number of minutes to
record.
• Recorded Sound Variable Name. Select a previously defined sound variable, or
enter the name of the new variable.
Play Recorded Sound
This action plays a previously recorded sound to the caller.
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• Recorded Sound to Playback. Select a sound from the dropdown. The dropdown
includes all sound variables that have been previously set in this IVR.
Send an Email
This action sends an email message. You can include IVR and System Variables in the
message, and you can attach a sound file.
• Email Address to Send To. Select a variable that contains the email address, or
enter an address in the text box.
• From Email Address. Select a variable that contains the email address, or enter
an address in the text box.
• Email Subject. Enter a subject in the text box. You can use IVR and System Variables to include text that is specific to this call.
• Email Body. Enter the text for the body of the email message. You can use IVR
and System Variables to include text that is specific to this call.
• Sound Attachments. Select a sound from the dropdown, then click Attach Sound
to Email. The sound name is displayed, with a remove button.
Send Recorded Sound to Voicemail
This action sends a sound to a folder in an extension owner’s voicemail Mailbox.
• Extension To Send Sound To. Select a previously defined variable, or enter an
extension in the text field.
• Variable of Sound to Send. Select a previously set sound variable.
• Voicemail Folder to Place Sound In. Select the voicemail Mailbox folder that
you want to send the sound to.
Store Recorded Sound in Sound Manager
This action stores a sound in the Sound Manager. This overwrites an existing user
sound file in the Sound Manager (it cannot overwrite a system sound). This action can
be used to re-record stored sounds, such as information updates.
CAUTION: When the IVR reaches this action, it immediately stores the chosen
sound in the Sound Manager, overwriting the existing sound. You probably want
to do something to limit the access to this action (i.e., require a pass code first).
• Folder. Select the Sound Manager folder that the sound should be stored in.
• Language. Select the Sound Manager language that the sound should be stored in.
• Sound to Store as. Select the Sound Manager sound file that should be over-written. This dropdown only includes the sounds available in the folder and language
that you select. You cannot select system sounds.
• Recorded Sound to Store. Select a previously defined sound variable that should
be stored in the Sound Manager as specified above.
Upload Recorded Sound
This action uploads a sound to a remote Web server. Two things are required: the URL
of the remote upload processing script, and an existing IVR sound variable.
You can enter a ‘static’ URL that is always the same, or you can create a ‘dynamic’
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URL by entering variables so that the system builds the appropriate URL.
A dynamic URL can include IVR Action variables or system variables. When using a
variable, make sure that a previous action has assigned a value to that variable,
otherwise nothing will be inserted into the URL. To add a variable to the URL, select
it from the appropriate Possible Variables dropdown, then click Append. The variable
will appear in the URL field. When the URL is requested, each
%VARIABLE_NAME% is replaced with the value of that variable.
• URL. Enter the URL to the remote processing script. Use the Possible Variables
options to append variables to the URL.
• Sound to Send. Select a previously defined sound variable from the dropdown.
• Timeout in Seconds. Select the number of seconds to try connecting to the URL.
If Switchvox cannot connect with the remote script within that time, Switchvox
will quit trying and go on to the next action in this IVR.
Merge Sounds
This action merges multiple sounds together into one sound variable.
• Sounds To Merge. Use the Existing Sounds and Sound Variables options to put
sounds into the boxes numbered 1 through 10. The sounds are merged in this
order.
• Variable Name for Merged Sound. Enter a unique name for this sound variable
(one sound that includes all of the above sounds merged).
Play Sound From URL
This action downloads a WAV sound file from a URL, then plays the sound.
You can enter a “static” URL that is always the same, or you can create a “dynamic”
URL by entering variables so that the system builds the appropriate URL.
A dynamic URL can include IVR Action variables or system variables. When using a
variable, make sure that a previous action has assigned a value to that variable,
otherwise nothing will be inserted into the URL. To add a variable to the URL, select
it from the appropriate Possible Variables dropdown, then click Append. The variable
will appear in the URL field. When the URL is requested, each
%VARIABLE_NAME% is replaced with the value of that variable.
• URL. Enter the URL to the remote processing script. Use the Possible Variables
options to append variables to the URL.
• Timeout in Seconds. Select the number of seconds to try connecting to the URL.
If Switchvox cannot connect with the remote script within that time, Switchvox
will quit trying and go on to the next action in this IVR.
• This sound is interruptible. Check this box if this is interruptible. For details, see
Interruptible Actions.
Gate Keeper
This action places a ‘gatekeeper’ in the IVR that counts how many times a caller has
traveled through the gate. This is useful to keep callers from getting stuck in infinite
loops, or to trap callers that are trying to abuse the system. Use a Conditional Clause
action to check the count, and send the caller to another IVR when they exceed the
limit.
• Gate Variable Name. Enter a unique name for this gatekeeper.
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Conditional Clause
This action is a conditional statement used to send callers to another IVR menu if the
statement is true. A Conditional Clause action has three parts: the variable to evaluate,
the criteria to use, and a value to compare against.
If your variable contains text, you can only use the criteria Equals or Does Not Equal.
The other criteria are only for use with a variable that contains numbers. Also, the text
you enter is case sensitive, so ‘RINGING’ does not equal ‘ringing.’
If your variable contains numbers, you can use any of the criteria.
• If. Select the variable to evaluate, the criteria to use, and a value to compare
against.
• Then Go To IVR Menu. Select the IVR menu the caller should be sent to if the
above statement is true.
• At Entry Point. Select an action that is in the above IVR menu. You do not have
to start at the beginning of the IVR.
Time Based Clause
This action checks the current date, day, and time and matches it against the Time
Frame. If the current time is within the Time Frame, then the caller is sent to another
IVR. See “Time Frames” ( page 80) for instructions on creating and modifying Time
Frames.)
For example, if the current time is within your Time Frame Outside Business Hours,
you can send callers to the IVR that plays a sound asking the caller to leave a message
or call back during business hours.
• If the current time is within the time range. Select the Time Frame to compare
against.
• Then Go To IVR Menu. Select the IVR menu the caller should be sent to if the
above statement is true.
• At Entry Point. Select an action that is in the above IVR menu. You do not have
to start at the beginning of the IVR.
Change Language
This action changes the ‘Call Language’ of this call, and any sounds played after this
action are played in this language (or the default language if there is no sound
available for the call language).
• Language to Change To. Select a language from the dropdown. The dropdown
includes the languages that are available in the Sound Manager.
Alter Caller ID
This action alters the caller’s callerID name or number. You can Prepend, Append, or
Replace the name or number with your information. This is useful to include
information that a caller has entered, or to show which IVR options the caller had
selected.
• How do you want to modify the Caller ID? Select Prepend, Append, or Replace
the callerID with your information.
• Which part of the Caller ID do you want to modify? Select Number or Name,
depending on which part you want to modify.
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• Use this text to modify the Caller ID. Enter the text that you want, or use the text
from a previously set variable.
Wait
This action adds a delay before the next action.
• Wait time in seconds. Enter the number of seconds to wait before continuing to
the next action.
• This wait is interruptible. Check this box if this is interruptible. See “Interruptible Actions” on page 68.
Give Busy Signal
This action gives the caller a busy signal for the number of seconds specified. The
busy signal is not interruptible, but it does not hang up the call. After it plays, the IVR
continues as normal.
• Signal Timeout in Seconds. Select the number of seconds that the busy signal
plays.
Hang Up
This action hangs up the call. There are no settings for this action, just click Save
Settings to save this action and go back to the Current Actions page.
Dial Extension
This action dials an extension. The caller leaves the IVR and rings that extension. At
that point, the extension’s Call Rules apply.
• Extension to Dial. Select a previously set variable, or enter an extension.
Send to Voicemail
This action sends the caller to the voicemail box of an extension.
• Extension to Dial. Select a previously set variable, or enter an extension.
Send to External Number
This action sends a caller to an external number.
• Number To Dial. Select a previously set variable, or enter an external phone number.
• Outgoing call rules for this number. Check the boxes to allow or deny use of
each outgoing call rule (just as you would for Switchvox extensions). See Outgoing Call Rules for details.
WARNING: Use this action with extreme caution because it can be abused by
callers.
Go To IVR Menu / Action
This action sends a caller to another IVR menu. This can also be used to jump or
return to an action in the current IVR menu.
• Go To IVR Menu. Select the IVR menu to send the caller to.
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• IVR Menu Entry Point. Select an Action that is in the IVR menu. (You don’t
have to start at the beginning of the menu.)
Record Digits
This action records digits (the DTMF tones) pressed by the caller. The digits are saved
in a variable.
• Beep before recording. Select Yes if you want a ‘beep’ sound to play before
recording begins. This lets the caller know they should start pressing buttons.
• Stop recording after this many digits. Enter a number from 0 to 255. Recording
stops when the caller has pressed the maximum number of digits, or the pound (#)
key. If this is 0, the recording continues until the caller presses #, or presses 255
digits.
• Recorded Digits Variable Name. Select a previously set variable, or enter a
unique name for this variable.
Say Digits/Letters
This action plays digits and/or letters to the caller.
Each digit or letter is read individually. For example, “one four four five,” or “L I K
E.” If you want to say “one thousand four hundred and forty five” instead, use the Say
A Number action. (If you want to say words, use a sound variable.)
• Digits to Say. Select a variable that contains the digits and/or letters to say, or
enter the digits and/or letters in the text field.
Say A Number
This action plays a number to the caller. The number 1445 is read as “one thousand
four hundred and forty five.” If you want it to say “one four four five,” use the Say
Digits/Letters action.
• Number to Say. Select a variable that contains the number to say, or enter the
number in the field.
Say Date/Time
This action plays a date and time to the caller. This can be the current date and time, a
specific date and time, or the date and time contained in a variable.
• Say the current date and time when the call enters this action. Check the box
Yes if you want this to be true.
• Say a predefined date. Enter the date to be played. Enter a time also, or leave it
blank.
• Say date from a variable. Select a variable from the dropdown. This variable
must be in epoch time.
• Format to Say the Date in. Select a format for the date to be played.
Play DTMF Tones
This action plays DTMF tones.
• DTMF Digits to Play. Select a variable that contains each of the DTMF digits to
play, or enter the digits in the field.
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• Play each digit for this many milliseconds. Select the number of milliseconds
that you want each DTMF tone to play.
• Wait this many milliseconds between digits. Select the number of milliseconds
that you want to wait before the next tone is played.
Perform Math
This action performs a mathematical operation on two IVR variables, and stores the
result in a new variable.
• Choose the variables for your operands and the desired operation. Select the
first variable, the operation, and the second variable.
• Variable Name for Result. Enter a unique name for this variable.
Set Variable
This action sets a variable.. IVR Variables are temporary variables that are set during
a call, and are only available to that call. During the call, the variable and its value can
be accessed from all IVR menus.
NOTE: If you want to set an IVR Variable where the value can be accessed
across calls, see Set Global Variable.
• Variable to set. Select a previously existing variable from the dropdown, or enter
a unique name in the text field.
• Set to value of. Select a System Variable, or enter a value in the field. If you select
a System Variable, the value of that variable is now contained in this variable.
Set Global Variable
This action sets a Global Variable.
Global Variables are variables that can be accessed from all IVR Menus and whose
values persist between calls. If a Global Variable’s value is changed by an IVR Menu,
all IVR Menus and all calls will have access to that value.
• Global Variable to Set. Select a Global Variable from the dropdown, or enter a
unique name for this Global Variable.
• Set to value of Variable. Select a variable from the dropdown. The value contained in the variable is now also contained in this Global Variable.
Get Global Variable
This action stores the value of a Global Variable into a normal variable. If you change
the value of this normal variable, it does not affect the original Global Variable.
Global Variables are variables that can be accessed from all IVR Menus and whose
values persist between calls. If a Global
Variable’s value is changed by an IVR Menu, all IVR Menus and all calls will have
access to that value.
• Variable to set. Select a previously existing variable from the dropdown, or enter
a unique name in the text field.
• Set to value of Global Variable. Select the Global Variable that contains the value
that you want.
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Concatenate Variables
This action concatenates multiple variables together, then saves the result in a new
variable.
• Variable Concatenation. Enter a representation of the variable you want to create. Add variables to this field using the Possible Variables options, or enter text.
Variables must be enclosed with %.
• Variable Name for Concatenation. Enter a unique name for this variable.
Send Call Values to URL
This action passes call values to the URL of a processing script. That script can return
XML containing a set of values, which Switchvox can save in variables.
• URL. The URL can include IVR variables or System variables. When using an
IVR variable, make sure that a previous action has assigned a value to that variable, otherwise nothing will be inserted into the URL. To add a variable to the
URL, select it from the appropriate Possible Variables dropdown, then click
Append. The variable will appear in the URL field. When the URL is requested,
each %VARIABLE_NAME% is replaced with the value of that variable. For
example:
http://www.mybiz.com/cgi-bin/
app.cgi?act=%ACCOUNT%&pin=%PIN%
This script might verify the account number and pin, then return a set of XML
that includes the account information.
• XML and Returned Values. If your script returns a set of XML with values that
you want to save and use, you must enter the variable names in the boxes labeled 1
through 8. After Switchvox sends the call values to your script, and collects the
return set of variables, those variables and their values are available for use on this
call, in this IVR menu.
The following is a sample DTD for the XML response from your script:
<!-- Switchvox PBX IVR xml DTD -->
<?xml version="1.0"?>
<!ELEMENT ivr_info (variable+)>
<!ELEMENT variable (#PCDATA)
<!ATTLIST variable name CDATA #REQUIRED>
The following is a sample XML response from your script:
<?xml version="1.0" encoding="ISO-8859-1"?>
<!DOCTYPE ivr_info SYSTEM "http://www.mybz.com/xml/ivr.dtd">
<response>
<result>
<ivr_info>
<variables>
<variable>
<name>balance</name>
<value>12323</value>
</variable>
<variable>
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<name>first_name</name>
<value>John</value>
</variable>
</variables>
</ivr_info>
</result>
</response>
Get Extension Status
This action gets the status of the extension, and saves the result in a new variable. This
is useful if you want to keep control of the call. For example, if you use ‘Get
Extension Status’ and the extension is busy, you could try someone else. If you used
‘Dial Extension’ and the extension is busy, Call Rules might send the caller to
voicemail instead.
Get Extension Status returns one of the following results:
•
•
•
•
•
•
IDLE
INUSE
BUSY
INVALID
UNAVAILABLE
RINGING
These are case sensitive, so IDLE does not equal idle.
• Extension To Check Status On. Select a variable that contains an extension, or
enter an extension number in the field. Click on the people icon to look up the
extension that you want.
• Variable Name for Status Result. Enter a unique name for this variable.
Get Extension Type
This action gets the type of the extension, and saves the result in a new variable.
Get Extension Type returns one of the following results:
•
•
•
•
•
•
•
•
•
•
•
•
•
zap
virtual
ivr
sip
conference
group_pickup
intercom
voicemail
callpark
agentcallbacklogoff
call_queue
agentcallbacklogin
autodial
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• directory
• simple_conference
• feature_code
These are case sensitive, so ZAP does not equal zap.
• Extension To Get Type Of. Select a variable that contains an extension, or enter
an extension number in the field. Click the people icon to look up the extension
that you want.
• Variable Name for Result. Enter a unique name for this variable.
Check User Password
This action evaluates the extension’s password with the password contained in a
variable, and saves the result in a new variable. This action returns a 0 (zero), which
indicates failure, or 1 (one), which indicates success.
• Choose variable where the extension is stored. Select the variable that contains
the extension.
• Choose variable where the password is stored. Select the variable that contains
the password.
• Variable Name for Result. Enter a unique name for this variable.
Set Distinctive Ring Hint
A Distinctive Ring Hint is used by an extension's Distinctive Ring Rules. These let an
extension-owner set up different 'Ring Actions' depending on the call.
To set a distinctive ring hint:
• Use the value of an existing variable. The hint is the value of the variable, and this
value must be previously set in the IVR actions. Otherwise, the hint is not available to the recipient of the call.
• Enter a value in the text field. The hint is exactly what you enter in this field.
NOTE: Distinctive Ring Rules are only available on Polycom phones.
Remove Distinctive Ring Hint
A Distinctive Ring Hint is used by an extension's Distinctive Ring Rules. These let an
extension-owner set up different 'Ring Actions' depending on the call.
To remove a distinctive ring hint:
• Use Options are listed at the bottom, with the folder icon. the value of an existing
variable. Select a variable from the dropdown. The hint that is removed is the
value of this variable.
• Select an existing Ring Hint from the dropdown. Select a Hint from the dropdown.
If you do not have any IVR actions of Set Distinctive Ringtone Hint, this dropdown is not available.
• Clear all Distinctive Ring Hints. Check this box if you want to remove all distinctive ring hints that were previously set in this IVR.
NOTE: Distinctive Ring Rules are only available on Polycom phones.
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IVR Options
IVR Options are entry points into another IVR Menu. In the IVR Tree, they are
shown as folder icons. Clicking the plus sign (+) next to an Option folder expands and
displays the actions in that IVR Menu. Clicking a folder displays it in the IVR Menu
Details. Clicking any of the Actions in the folder displays that action in the IVR Menu
Details.
To create a new IVR Option, click the Create Option button under Options in the
IVR Menu Details. The following fields are used when creating an IVR Option.
Option Number. Select a number, or the * key. This what the caller presses to select
this option.
New IVR Menu. Select from the dropdown menu the IVR menu the caller is sent to
after the Option Number is pressed.
IVR Menu Entry Point. Select from the dropdown menu an Action that is in the IVR
menu. (You don’t have to start at the beginning of the menu.) Click the Save Option
button. This displays the Modify Option page, showing the Current Options. Select
or enter the following, click any of the Save buttons when finished.
• Enable Extension Dialing. Select an Extension Group from the dropdown menu
that includes the extensions you want to allow callers to dial during this IVR. For
example, the IVR might play a sound “if you know your party’s extension, please
dial it now.” You can also leave the setting at the default No extension dialing, if
you wish.
Normally IVR Menus only allow single digit options. However, if this field is set,
then IVR callers (in this IVR Menu only) can dial extensions. If the IVR caller
enters a valid extension, Switchvox rings the extension. If the caller enters an
invalid extension, then it is considered an “invalid option,” and is handled
according to that setting.
Options and extension dialing can coexist in the same IVR Menu. However, if
you have an option in the menu (i.e.; 1) that starts with the same number as an
extension (i.e.; 105) then the caller experiences a delay. When he presses ‘1’,
Switchvox waits a moment to see whether he might also press ‘05’. If caller
doesn’t, Switchvox proceeds to option 1.
• Invalid Option Settings. Specify what happens when a caller presses a key for an
option that doesn't exist. Select a sound to play, then an IVR menu and entry point
(you don’t have to start from the beginning of the menu).
• Option Timeout Settings. Specify what happens if a caller fails to select any
options within a specified amount of time. Set the number of seconds to give the
caller, a sound to play, and an IVR menu and entry point (you don’t have to start
from the beginning of the menu).
• Retry Settings. Specify what happens if a caller chooses an invalid option or
times out several times. Select the number of times to allow, an IVR menu, and
entry point (you don’t have to start from the beginning of the menu).
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Time Frames
Switchvox can operate differently based on the date, day, and time. You can set up
IVR Menus and Call Rules to check the Time Frames, and then act accordingly.
There are several predefined time frames that you can use: Business Hours, Outside
Business Hours, and Weekends Only. You can modify or delete these time frames, or
create new ones.
IMPORTANT: A time frame is considered valid when the current date and time
match any one of a time frame’s conditions. Not all of the conditions in the time
frame have to match the current date and time.
Creating a New Time
Frame
Time frames consist of Time Frame Conditions. The time frame itself is considered
valid when any of its conditions are met. To modify a Time Frame with Conditions:
• Click Create Time Condition.
• Enter a Date Range, Day of the Week Range, or a Time Range, or any combination.
• Continue adding conditions until you have completed the Time Frame definition.
The following are sample time frame definitions:
Start Date
End Date
Start Day
End Day
Start Time
End Time
-
-
Monday
Monday
-
-
-
-
Wednesday
Wednesday
-
-
Mondays 12:00am to 11:59pm, and Wednesdays 12:00am to 11:59pm
Start Date
End Date
Start Day
End Day
Start Time
End Time
-
-
-
-
3:00AM
6:00AM
Start Time
End Time
Start Time
End Time
Early Morning
Start Date
End Date
07/01/2014
07/07/2014
Start Day
End Day
The First Week in July 20114
Start Date
End Date
Start Day
End Day
01/21/2014
After January 21, 2014
Start Date
End Date
Start Day
End Day
Start Time
End Time
02/01/2014
02/28/2014
Monday
Friday
6:00AM
7:00AM
Weekdays in February, 2014 between 6:00 and 7:00 in the morning
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Modifying a Time
Frame
• Click its Modify icon. This displays the Modify Time Frame Conditions page.
• Click a Delete icon to remove a condition that no longer applies. (You cannot edit
a condition.)
• Click Create Time Condition to add a condition.
Deleting a Time
Frame
To delete a time frame, click its Delete icon. Be sure that you do indeed want to delete
this time frame, then click Yes, Delete. The Time Frames main page is displayed with
a message that the time frame was deleted.
NOTE: Before you delete a Time Frame, be sure you do not have any IVR Menus
or Call Rules that depend on it.
Call Recording
Call Recording lets you set up rules so Switchvox automatically records calls. Any
calls that match these rules are recorded and listed on this page. You can view, listen
to, and download recorded calls. There is one default call recording rule that Switchvox automatically follows: user-initiated recordings. When a user records a call (his
own call or another person’s), those recordings are included here. These are the same
recordings the user has in his or her Mailbox, so if you or the user deletes one of these
recordings, it is no longer available in either area. However, if the user forwarded the
recording to another Switchvox user, that version is unchanged.
Scheduling a Call
Recording
To schedule a new call recording, click Create Call Recording. This displays the
Create Call Recording page.
General Settings
Call Recording Label. Enter a unique name for this scheduled call recording rule.
Recording Targets
Select one or more extensions, extension groups, or call queues. You cannot combine
these types, but you may select many of each type. Choose from one of the following
types:
• Phone Extensions. Search for and select the desired extensions.
• Call Queues. Search for and select the desired call queues.
• Extension Group. Search for and select the desired extension group.
Types of Calls to Record
YES or NO, select the types:
• All Outgoing Calls
• Incoming Direct Calls
• Incoming Queue Calls.
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Call Recording Limit
Set a specific number of calls, or the date and time you want the recordings to end.
How long to keep the recordings
Set the storage instructions for the recordings. If you want to back up older recordings
before you delete them, click the link to Call Recording Settings Section (see Call
Recording Settings).
IMPORTANT: If you enter a number into the Delete calls after setting, Switchvox immediately deletes any older recordings to ensure there are no more than
that number of files on disk.
Force Minimum Recording Length
Enable Minimum Recording Length. Select YES or NO.
Delete recordings shorter than. If minimum length is YES, enter a value in seconds.
Viewing & Playing
Recorded Calls
Click the Recorded Calls tab to view the Recorded Calls list, which shows the information for each recorded call and lets you play the call. When you click Play, you can
choose to play the call with your audio player, or download the file (these options may
depend on your Web browser’s settings).
Filter Options
Click the arrow to show filter options for viewing the calls. Filter options include the
following. Click Run Filter to limit the Recorded Calls to the calls that match your
filter.
All. Shows all recorded calls in your system.
Date. Enter the From Date and To Date for the time period from which you would
like to hear the recorded calls. Use the mm/dd/yyyy format.
Recording Tag. Choose the recording tag of the recorded calls that you would like to
hear from the dropdown menu.
Users. Search for and select the user you wish to hear calls from.
Call Queues. Search for and enter the queues you wish to hear calls from.
Extension Group. Search for and select the extension group you wish to hear calls
from.
Caller ID. Select either From or To for the Call Direction. Then enter the phone
number, extension, or caller ID of the person you wish to hear the recorded calls
Duration. Select the time limit of the recorded calls from the dropdown menu you
wish to hear from.
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Call Recording
Settings
You can back up your recorded calls to an external FTP server before the calls are
deleted by clicking Call Recording Settings, then entering the FTP server settings
and the offload/sync schedule. Click Save Settings when finished.
External Recording Settings
then entering your FTP server settings: FTP Hostname, FTP Username, FTP
Password, FTP Path, and Use SFTP. This is highly recommended if you rely on call
recordings. You can test the FTP Settings
When recorded calls are backed up to the specified FTP server, Switchvox
automatically creates a separate folder on the FTP server for each recording rule. Each
call recording is automatically stored in its respective folder.
Each recorded call has two files, a WAV file and a XML file. The WAV file is the
sound, and the XML is the data about the call. The file names are based on the
following structure: [Date]-[Time Stamp]-[From Caller ID]-[To Caller ID]. These are
samples:
2007-04-06-15-00-41_7771021212_102.wav
2007-04-06-15-00-41_7771021212_102.xml
This sample call was made on April 6, 2007 at 3:00:41PM. A call from 7771021212
was received by extension 102.
The XML file includes the following data:
<recording_info>
<recording_tag>Steve Smith</recording_tag>
<recorded_call_id>1200</recorded_call_id>
<recorder_cid></recorder_cid>
<recorded_cid>Steve Smith &lt;102&gt;</recorded_cid>
<recorder_account_id></recorder_account_id>
<recorded_account_id>1101</recorded_account_id>
<from_account_id>0</from_account_id>
<from_caller_id>BuyIt &lt;7771021212&gt;</from_caller_id>
<to_account_id>1101</to_account_id>
<to_caller_id>Steve Smith&lt;102&gt;</to_caller_id>
<duration>316</duration>
<date_created_ts>2007-04-06 15:00:41.611474</date_created_ts>
<date_created_secs>1175896842</date_created_secs>
</recording_info>
Offload/Sync Schedule
These settings control how often call recordings are offloaded to the FTP server. This
is also how often the user-initiated recordings seen here in the Scheduled Call Recordings are sync'd with the recordings in the phone-extension mailboxes.
This can be resource-intensive, so you may want to do it only once a day, outside of
business hours. Otherwise, Switchvox automatically does this throughout the day.
Deleting Recorded
Calls
To delete call recordings, check the box for each one that you want to delete, then
click Delete Checked (at the top of the list). Be sure that you do indeed want to delete
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the recordings, then click Yes, Delete. If you are deleting ‘user-initiated recordings,’
the recordings are deleted from the user’s Mailbox.
Digium Addon Products
This lets you register and enable Digium Addon products.
To add a product,
• Select the product from the dropdown.
• Enter your License Key.
• Click Submit.
NOTE: If you install a Fax license, you will be prompted to download and install
the related fax software.
Finding Product
Addons
For information about what Addons are available for your Switchvox, go to Switchvox Product Addons.
On the Switchvox Product Addons website, you must enter your Switchvox Registration Code to find your Addons. You can find your Registration Code on your Switchvox DVD.
Your Addons could include your Switchvox Fax License, other freely available
Addons, and other available Addons that are specific to your Switchvox.
Event Triggers
Event Triggers let you set up Switchvox to interact with external web-based applications. When certain events occur in Switchvox, it can do an HTTP POST that includes
information about the event, and then respond to XML that is returned. Click Save
Event Triggers when finished.
System Default URL
You can enter a Default URL so that you do not have to enter a full URL for each
event. But, you can choose to use the default URL or not, so you still have flexibility
on each event.
Event Specific URLs
There are some Switchvox events that trigger an HTTP POST. You can choose to
Append URL variables and to use the Default URL (YES/NO).
On Incoming Call Event URL. Triggered at the beginning of an incoming call. This
event applies to calls received from VOIP providers, any channel groups, as well as
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calls between extensions. It is common for requests of this type to return XML for
routing the call or altering the Caller ID.
On Route to Extension Event URL. Triggered when a call hits an extension. This
event can occur multiple times for each call. For example, a call may hit an IVR, a
queue, and then a phone extension. XML can be returned to route calls differently.
WARNING: Do not create routing loops. For example, if a call to extension 333
results in a request that sets transfer_extension to 333, then Switchvox loops endlessly. Do not set transfer_extension in the response to this event unless it is absolutely necessary.
On Call Answered Event URL. Triggered when a call is answered. This event
applies to calls from external numbers, and from all Switchvox extensions.
On New Voicemail Event URL. Triggered when a voicemail message is left for an
extension (including a message forwarded from another extension). This event
requests the voicemail URL.
NOTE: Setting XML values in the response to this request results in undefined
behavior. Return empty XML for this request.
On Checked Voicemail Event URL. Triggered when an extension checks its
voicemail. This event could be used in conjunction with the On New Voicemail Event
to track new voicemail messages and implement an escalation policy.
NOTE: Setting XML values in the response to this request results in undefined
behavior. Return empty XML for this request.
On Agent Logged In Event URL. Triggered when a queue member logs into a
queue.
On Agent Logged Out Event URL. Triggered when a queue member logs out of a
queue.
On Call Hangup Event URL. Triggered when a call is hung up. This event can be
used in conjunction with the JOB_ID variable to record the completion of calls that
invoked previous URL events. Note: Setting XML values in the response to this
request results in undefined behavior. Return empty XML for this request.
On Outgoing Call Event URL. Triggered when a call is created. This event applies
to calls to external numbers and all Switchvox extensions.
URL Variables
Switchvox events can submit various information about calls. To do this, use
variables in the URL. Not every variable is available in every event (check the
Acceptable Variables dropdown on each event), so be prepared to handle an empty
variable. Here is the complete list of all of the variables for all event types:
CALLER_ID_NAME. Name from caller ID.
CALLER_ID_NUMBER. Phone number from caller ID, including area code.
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DIALED_NUMBER . event type associated with outgoing call events.
EVENT_TYPE. Event type that has been triggered. This is useful for the Default
URL, which receives POSTs from any of the event types.
EXTENSION. Internal extension associated with the event.
EXTENSION_TYPE. The type of the extension associated with the event. For the
list of extension types, see Extension Types.
FEATURE_DATA. The data entered after a feature code. For example, in *62204,
you have an EXTENSION of 62, an EXTENSION_TYPE of feature_monitor, and a
FEATURE_DATA of 204.
INCOMING_DID. Phone number dialed by an external caller. May be empty for
internal calls, or calls over certain analog lines that do not provide the DID.
JOB_ID. Unique ID of the call. Note: This string of digits is not a numerical value:
99.100 is not equal to 99.1. This variable is useful for correlating multiple events (and
also certain IVR requests).
QUEUES. A comma separated list of the queue account_ids being logged in to or
logged out of. In the case where the user is being logged out of all queues, the list will
only contain the list of queues the user is a member of.
VM_DATE. The date and time of the voicemail message.
VM_DUR. The duration of the voicemail message, in seconds.
VM_MAILBOX. Extension associated with the event’s voicemail box.
VM_MSGNUM. The unique message number for this voicemail message.
XML Response
For some events, the remote web application can return XML that instructs Switchvox
to perform operations on the call. Here is a sample of an XML response:
<response>
<result>
<call_info>
<caller_id_number>8585551212</caller_id_number>
<caller_id_name>John Doe</caller_id_name>
<transfer_extension>850</transfer_extension>
<display_url>http://intranet.example.com/crm.cgi?custid=2</
display_url>
<continue_outgoing_reporting>1</continue_outgoing_reporting>
</call_info>
</result>
</response>
All of the tags inside the call_info block are optional. The following XML tags can be
used:
callerid_num. Set Caller ID Number to this value.
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callerid_name. Set Caller ID Name to this value.
transfer_extension. Transfer call to the extension of this value. If this XML is a
response to the event On Outgoing Call, this can be an external number. Make sure to
enter this number as you would dial it from your phone. For example, if you must dial
9 for an external number, an external number here must begin with a 9. WARNING: It
is possible to break Switchvox by creating routing loops with transfer_extension.
display_url. Display this URL in the Switchboard as a clickable icon. For example:
the remote web application might lookup the caller’s phone number in your
company’s user database, then return the URL of the caller’s user record.
continue_outgoing_reporting. Continue reporting any subsequent outgoing calls for
this JOB_ID. For example, if a call is made to a queue, the event will continue when a
queue-member's extension is dialed. The value of this tag must be 1 or 0.
Voicemail
The following tools allow you to set up voicemail routing, external voicemail, and
custome voicemail message length,
Voicemail Routing
Voicemail Routing lets you handle several voicemail-related actions.
• During a user’s voicemail greeting, callers may push the “*” key to log into the
current voicemail box. Select YES if you want this to be true. When the caller
presses “*,” he or she is prompted for their password.
• During a user’s voicemail greeting, callers may push the “0” key to exit and be
forwarded to extension []. Enter an extension if you want this to be true. Click the
search icon to select from a list of extensions. When the caller pushes “0,” he or
she has the option to accept and save their recording before being transferred to
the extension.
• If a user’s voicemail box is full, callers will be forwarded to extension []. Enter an
extension if you want this to be true. Click the search icon to select from a list of
extensions.
Click Save Voicemail Routing Settings when finished.
External Voicemail
If you want to use the voicemail system of an external provider, rather than this
Switchvox’s voicemail system, select YES and choose from the dropdown menu the
SIP Provider whose voicemail system you want to use. Click Save External Voicemail Settings when finished.
In Switchvox SMB, the SIP Provider can be a peered Switchvox (for more information, see Adding a Peered PBX). To use a peered Switchvox as the External Voicemail
Provider: Be sure both Switchvoxes are running SMB version 4.0 or later.
On the server Switchvox (the External Voicemail Provider):
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• Set up a new SIP provider that is the client Switchvox. On the new provider, show
Advanced Options, select Yes for Host is a Switchvox PBX, and enter the Jabber
Hostname for the client Switchvox.
• Be sure that each extension that exists on the client Switchvox also exists on the
server Switchvox as a Virtual extension. For help managing this, see Import
Extensions.
On the client Switchvox:
• Set up a SIP provider that is the server Switchvox.
• Check the box to Use External Voicemail System, and select the server Switchvox
as the SIP provider.
• Set up an outgoing call rule so that calls to the voicemail-access extension (by
default, this is 899) are sent to the server Switchvox. For details, see “Outgoing
Call Rules (tab)” ( page 16).
NOTE: When checking their voicemail, users should go to the Switchvox User
Suite on the server Switchvox.
Custom Voicemail
Message Length
• You can specify the number of seconds that it takes for Switchvox to “give up” on
a caller if they reach your voicemail but don’t leave a message. For example: you
specify 10 seconds, someone calls and reaches your voicemail, but they don’t start
talking, then Switchvox hangs up and does not save those 10 seconds of silence.
Setting a number of seconds here should help reduce clutter in Mailboxes and on
disk.
• You can also set a number of seconds to let the caller speak, and after that the call
is disconnected. This is useful if you are concerned about storage capacity.
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Notification
Notification
SMTP Settings 87
Templates 87
SMTP Settings
When voicemail is received for a phone-type extension, a notification email message
is sent to the extension’s email address. SMTP Settings let you set a “From” address
for those email messages and the SMTP server. You may need to set this field to be a
real email address before you are able to receive voicemail notification emails. Some
strict mail servers bounce email that comes from the localhost.localdomain domain.
Server Settings
Voicemail Notification Email “From” Address
Enter an email address. When you receive emails notifying you of new voicemail
messages, they will come from this email address. You may need to set this field to be
a real email address before you are able to receive voicemail notification emails.
Some strict mail servers will bounce emails appearing from the domain localhost.localdomain.
Outbound SMTP Server
Enter the hostname of your SMTP server. Some ISPs or network admins require all
outgoing email to pass through a specific SMTP server. If you are having trouble
receiving voicemail notification emails, you may want to specify the hostname of
your server here.
Outbound SMTP Username and Password
Enter a username and password if your SMTP server requires it. It’s probably safe to
leave these blank, as many mail servers do not require them.
Notification
Diagnostics
If you are having problems with voicemail notifications, enter an email address to
send a test message, then click Run Diagnostic Tool. A diagnostic page is displayed
that describes each step that was taken, and the result (passed or failed).
Notification Templates
Notification Templates let you customize voicemail notifications. You can create
multiple templates and tailor them to different types of accounts. These templates are
available to extension owners when they set up voicemail notifications (along with
any templates created).
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Notification
You can use variables in your template that are substituted with real values when the
notification is sent. For example, %VM_NAME% in a template sends as the extension owner’s first and last name. There is a full list of these variables on the pages
where you create or edit a template.
Create Notification
Templates
After you select Templates from Tools >Notification menu, the Notification Templates page is displayed. This page lists the existing templates, including the default
templates, the date they were created, along with the available Actions.
NOTE: You cannot delete a default template.
To create a new template, click the Create Notification Template, which displays the
Create Notification Template page with default settings that you can change if you
wish. You can modify the subject and body (change it completely if you need to),
using the list of possible variables at the bottom of the page. You can leave the subject
blank, but you must put something in the body of the message.
Template Settings
Template Name. Enter a name in this field.
Subject. Enter the subject. You can use variables from the Available Template
Variables to personalize it.
Body. In the message box, enter your message, using variables from the Available
Template Variables to personalize it.
Click Save Notification Template to save your changes and go back to the Notification Templates page.
Available Template Variables
%VM_NAME% - Recipient's firstname and lastname
%VM_DUR% - The duration of the voicemail message
%VM_MAILBOX% - The recipient's extension
%VM_CALLERID% - The caller ID of the person who left the message
%VM_DATE% - The date and time the message was left
%DOWNLOAD_LINK% - A URL to download the message
%DOWNLOAD_LINK_AND_MARK_DELETED% - A URL to download the message
and delete it from the recipient's mailbox.
Modify a Notification
Template
From the list on the Notification Templates page, click the modify icon for the template you wish to modify. The Modify Notification Template page is displayed.
Click Save Notification Template to save your changes and go back to the Notification Templates page. Click Restore to Original Template to restore template to the
original system default.
To see a preview of a template, click its View icon. The template is displayed in both
HTML and text-only.
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Extension Features
Deleting Notification
Templates
You can edit each of the system default templates, but you cannot delete them.
To delete one of your own custom templates, click its Delete icon. Be sure you do
want to delete the template, then click Yes, Delete. The template is deleted, and the
Notification Templates page is refreshed with the deletion removed. If any extension
owners were using this template, those settings are changed to the Default template.
Extension Features
Contacts 89
Distinctive Ringtones 90
Switchboard Panels 90
Digium Phone Apps 93
Contacts
Internal Directory
By default, the Internal Directory is available on all of your Digium phones, and it
includes all of the Switchvox extensions. (The Internal Directory is also available on
Polycom phones configured with a Phone Feature Pack, under the Switchvox Directory application.)
You can customize what extensions are in the Internal Directory by building one or
more Extension Groups and adding them to the directory. If you do this, be sure to
remove the existing group, All Extensions.
Download the
Internal Directory
Click the floppy disk icon to download the XML for the Internal Directory. This is
useful for a peered Switchvox to upload as an External Contacts file.
External Contacts for
Digium Phones
To add external Contacts (Contacts that aren't extensions on this Switchvox) to all
Digium phones, you can upload a CSV or XML file.
To see an example of the XML used by Contacts, download the Internal Directory's
XML file.
CSV headers:
first_name, last_name, organization, job_title, location,
email_address, primary_number, primary_number_label,
secondary_number, secondary_number_label, jabber_id,
subscription_url
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Extension Features
Example of the CSV record for one of your vendors:
Jane,Doe,Bunches of Flowers,Designer,San
Diego,[email protected],918585551212,Flower
Stand,918585553333,Mobile,,,
Distinctive Ringtones
Switchvox can store and play distinctive ringtones that can be used on Digium Phones
(or a Polycom phone with a Phone Feature Pack). See Phone Features
There are limitations for the total number of ringtones, and the total amount of diskspace that can be used. But you should have plenty of room to add the ringtones that
you like.
To create a ringtone:
• Click Create Ringtone.
• Click in the Choose a File text box, and browse your computer to find a .wav
sound file.
• In the Name field, enter a name for this ringtone.
• Click Save Ringtone.
The Ringtones list is refreshed, with your new ringtone listed.
Distinctive Ringtones
List
This list shows the ringtones that you have uploaded to Switchvox, and the ringtones
that come with a Digium Phone (or a Polycom phone). You can do the following:
Play the ringtone. The audio file is downloaded to your computer, then played using
your default audio player for .WAV files.
Modify the ringtone. You can upload a new file, or change the Name.
Delete the ringtone. You must verify that you want the ringtone deleted. If you
delete a ringtone, every Ring-Rule that uses that ringtone is also deleted.
Switchboard Panels
Switchboard Panels let you add powerful panels to your Switchboard. These panels
let you exchange data with external Web applications, then display the results. The
panels refresh each time a call comes in, showing caller-specific information.
For each panel, you can specify which phone-type extensions are allowed to use it. If
an extension isn’t allowed to use a panel, that panel isn’t included in the Switchboard’s Panel menu.
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Extension Features
Standard Switchboard
Panels tab
Standard panels provide integration with SugarCRM®, Salesforce.com®, and
Google® Maps. To set up, click the Setup button for the desired panel. and when finished, click the Save Switchboard Panel button.
SugarCRM® Switchboard Panel
This panel provides integration with your SugarCRM applicationm displaying information from your SugarCRM® database about the incoming caller.
SugarCRMSettings
• SugarCRM Web URL. Enter the URL to your SugarCRM application’s Web
interface (e.g., http://crm.example.com/sugarcrm/). If you want external users to
have access to this panel, then this URL must be accessible from the outside Internet. If your SugarCRM application is running behind NAT, you may need to make
it accessible by forwarding ports. HTTPS URLs and alternate ports are supported
(e.g., https://public.example.com:8080/sugarcrm/).
• SugarCRM SOAP URL. Enter the URL to your SugarCRM application’s SOAP
API (e.g., http://crm.example.com/sugarcrm/soap.php). This URL must be accessible from the outside Internet, so if your CRM application is running behind NAT,
you may need to make it accessible by forwarding ports. HTTPS URLs and alternate ports are supported (e.g., https://public.example.com:8080/sugarcrm/
soap.php).
• Extensions allowed to use this panel. Search for and select one or more extensions to allow them to use this panel. All extensions displayed in the box are
allowed to use this panel.
Test CRM Settings. Click this button, then enter your CRM system username and
password, and the phone number for a contact in your application. Click Test Settings
to see the results of your test. This section is for testing your Switchvox's access to the
CRM service, this Username and Password are not used outside of this test. In the
Switchboard panel, users must enter their own login information.
Salesforce.com® Switchboard Panel
This panel provides integration with your Salesforce.com account whenever an
incoming call is received, showing the caller's information from your Salesforce.com
database.
Salesforce Settings
• Salesforce Web URL. https://na1.salesforce.com has been entered as default.
• Extensions allowed to use this panel. Search for and select one or more extensions to allow them to use this panel.
Test CRM Settings. Click this button and enter your CRM system username and
password, and the phone number for a contact in your application. Click Test Settings
to see the results of your test. This section is for testing your Switchvox's access to the
CRM service, this Username and Password are not used outside of this test. In the
Switchboard panel, users must enter their own login information.
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Extension Features
Google® Maps Switchboard Panel
This panel provides integration with Google Maps to pinpoint the location of incoming calls, based on their caller ID.
Google Maps Settings
• Default Zoom Level. Select the level of zoom to use. The Switchboard user can
always zoom in or out.
• Extensions allowed to use this panel. Select one or more extensions allowed to
use this panel.
Test Google Map Lookup. Click this button and enter a phone number to look up.
Click Run Test to see the results of your test.
Custom Switchboard
Panels tab
Custom panels let you build your own panel that integrates with any web application
you choose. You can build the URL with Switchvox variables, which pass caller data
to the application. Click Create Custom Panel and fill in the following:
Panel Settings
Panel Name. Enter a unique name for this panel. The panel name appears as the title
of the panel window.
Panel Description. Enter a description of this panel. This is only displayed in the
Admin Suite, it is not shown in the panel.
Panel URL. This is the URL to be loaded in the panel, for example:
http://news.google.com/news?q=%CALLER_ID_NAME%
The URL can contain variables (between percent signs) that are substituted with real
values during each request. (See URL Variables.)
URL Variables. You can include one or more variables in a Custom Panel URL.
CALLER_ID_NUMBER
The phone number of the current caller. Example: 8585551212
CALLER_ID_NAME
The text component of Caller-ID for the current
caller. Example: BOB JONES
EXTENSION
The extension of the current Switchboard user.
This does not apply to who is calling, but is
instead the extension who has launched this
Switchboard panel.
SESSION_ID
A string of numbers and underscores representing the current switchboard session. (There is no
guarantee that this session_id is unique across
logins or machines.)
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Extension Features
FIRST_LOAD
A boolean integer value indicating if this is the
first time the panel is requesting the page. On
startup, the panel will request the URL without
any associated call information with
FIRST_LOAD set to 1. After subsequent calls,
FIRST_LOAD will be set to 0.
PARTIAL_REG_CODE
The last four digits of your Switchvox’s registration code.
FULL_REG_CODE_DIGEST
A hexadecimal MD5 hash of your Switchvox’s
registration code.
Extensions allowed to use this panel. Search for and select one or more extensions
allowed to use this panel.
Click Save Switchboard Panel when finished.
Digium Phone Apps
The Digium Phone API lets you build custom apps for your phones using the popular
programming language JavaScript. For more information about writing Digium
Phone Apps, see the Digium Phone API Information Center at phones.digium.com.
After you have written your phone app, use this tool to deploy the app to your Digium
Phones.
Create App
To upload an app for your Phones, click the Create App button. This displays the
Upload Digium Phone App screen from which you specify General Settings information, Extensions that Use the App, and Configuration Settings. After you have
entered this information, clicking Save App uploads the packaged file containing the
app and its associated files.
General Settings
Packaged File. Click the text box to display the contents of your hard drive. Select
the app file, which must be a zip file containing the source files that define the app.
Note. A text box is supplied in which you may enter optional notes about the app.
Always Running switch may be set to YES so the app will be started if it is not
running. NO does not start the app.
Include API Credentials switch may be set to YES so the extensions’ credentials are
included. This is only necessary if you are using pbx.request and Switchvox Extend
API methods.
Extensions that Use App
This option allows you to indicate which Switchvox extensions are to have the app.
Use the search box to filter the list of potential extensions. Highlight the extension to
move it to the table. To remove an extension, highlight the entry in the table and click
the red removal X.
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Media
Configuration Settings
If your app uses configuration settings, you can include them here.
If you are deploying this app to multiple extensions and need to use extension data as
a value, use the Insert an Extension Setting Variable button.
Use the Add Setting button to add more fields to the table, and use the Clear All
Settings button to clear all of the settings.
When finished, click Save App. This saves the app in Switchvox, and deploys the app
to the phones as indicated..
Digium Phone Apps
All installed apps are listed on the Digium Phone Apps main screen in alphabetic
order. To filter the list, you can enter a character in the Type to Search box. The list
indicates the app Name, if the app is Always Running, and if it Has Idle Screen.
There are four Action buttons for each app also: Display of its Details, a Modify button that displays the same options used when creating its entry, a Download button to
save the app package to your hard drive, and a Delete button to remove the app from
the phones and Switchvox.
Media
Music On Hold 94
Sound Manager 96
Music On Hold
The Music On Hold (MOH) tool allows you set up audio programs for callers to listen
to when they are not actively on a call (i.e., on hold, waiting in a queue, parked, or
being transferred). Although audio files in MOH are referred to as “songs,” such entities may also be custom announcements that can be updated as often as you like as
long as they are in MP3 or Ogg Vorbis codecs. This means any file you want to use for
MOH needs to end in .mp3 or .ogg, or it won’t play.
The MOH tool has two tabs: Songs and Groups. Songs lists songs and groups of songs
and provides a means to upload songs to already created groups. Groups provides a
means to create a new group and to make group settings. Most MOH situations use
the default group, which you can customize by adding and deleting songs. MOH in a
call queue can use any group. For example, you can create a MOH group with
announcements about special offers and assign that group to your sales call queue.
Songs tab
Upload Song
Click this button to display the Song Information popup.
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Media
• Song to Upload. Click inside the text box to display the contents of your hard
drive, then select a file to upload. The filename will be automatically placed in the
Song to Upload box and the Name box.
• Add to Group. If you want to add the song to an existing group, select the group
from the dropdown menu.
• Click the Upload Song button, which redisplays the Songs tab with the new song
in the list.
View
• View. Click the dropdown menu and select from the options what you wish to
view: All Songs, the default group listing, or any defined Group Listings. Your
selection is displayed in a table.
-All Songs includes a check box, the Name, Group Name, Play Order in its
group, and available Actions for each song.
-Default includes the current default group.
-Group listings include the selection in a table with a check box, Move arrow,
Name, Group Name, Play Order, and Actions available for each song.
• To sort by column data, click that column name, and click it again to toggle
ascending/descending order. You can modify a song name by clicking its pencil
icon or play a song by clicking its Play icon.
Moving or Copying Songs
To move song files to a different group, check the box for each song that you want to
move, select the new group from the Move Checked To (icon) dropdown, then click
the icon. The song list is refreshed and reflects this change.
To copy song files to a different group, check the box for each song that you want to
copy, select the new group from the Copy Checked To (icon) dropdown, then click the
icon. The song list is refreshed and reflects this change.
Deleting Songs
To delete song files, check the box for each song that you want to delete, then click
the Delete icon. This displays the Delete Music On Hold Files popup men. Review
the list of songs that will be deleted, be sure that you do indeed want to delete them,
then click Yes, Delete Files. The files are permanently deleted, the Music on Hold
page is redisplayed, and a success message indicates that the files were deleted.
Groups tab
Create Group
To add a new group, click the Create Group button.. This displays the Group
Information popup menu. Supply the following information:
•
•
•
•
•
Group Name. Enter a unique name for this group.
Description. Enter brief description of the group.
Play Volume. Select a percentage sound level on the adjustable slide bar.
Play Order
Select Chosen Order, which plays in the order displayed in the list, or Random,
which randomly plays all the songs in the list.
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• Click Save Group when finished. This redisplays the Music On Hold Groups
page with the new group listed in the table.
Group Settings
To set the Default System Group, click the Group Settings button. This displays the
Group Settings popup menu. Select the default system group from the dropdown
menu. When finished, click the Save Music On Hold Settings button.
Modifying a Group
Click the pencil icon for the group you wish to modify. This redisplays the Group
Information popup. Make your changes, then click Save Group to save your
changes and go back to the Music on Hold page. A success message indicates that the
group has been modified.
Deleting a Group
Click the delete icon for the group you wish to delete. This displays the Delete Music
on Hold Group popup menu. Be sure you really want to delete the group, then click
Yes, Delete Group. This redisplays the Music On Hold main page with an updated
group list.
Sound Manager
The Sound Manager lets you create, upload, modify, and delete all of the sounds in
Switchvox other than Music On Hold. It also lets you organize your sounds in folders,
and manage the same sounds that are in different languages. Sounds can be uploaded
from your computer, or recorded over any phone that’s connected to Switchvox.
From the main Sound Manager page you have View and Create options.
View offers several display options, which are shown in separate tables:
•
•
•
•
•
All Sounds
All Sound Packs,
All Folders
All Languages
All sounds within a selected folder
Create allows you to add a new Sound, Sound Pack, Folder, or Language.
NOTE: You are not allowed to delete, move, or modify system sound information or system sound files in the English language.
Adding a Sound
To add a new sound, select Sound from the Create dropdown menu and click Go.
This displays the Create Sound/Sound Information page. Supply the following
information:
Sound Name. Enter a unique name for this sound.
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Media
Sound Description/Script. Enter the words of this recording. This is displayed when
you mouse over the sound name in the list of sounds.
Folder. Select a folder from the dropdown menu for this sound.
Sound Language. Select the language that is used in this sound.
Sound Source. Select either Record Over Phone or Upload File from the dropdown
menu.
-If you chose Record Over Phone, enter the extension to ring, then click Ring
Phone. Make your recording, then hang up the phone or press #.
-If you chose Upload File, browse your computer to select and upload the sound
file.
NOTE: Only wav or gsm files are supported for sound files.
When the settings are correct, click Save Sound. (If you are uploading a file, the
transfer begins). The system redisplays the Sound Manager page with a message that
a new sound has been added, and the new sound is included in the list of sounds.
Modifying/Playing a
Sound
Click a file’s Modify/Play (pencil) icon to display the Sound Info page that shows
the sound’s Status, Language, Sound Information, and Actions allowed.
Play Sound Icon. Plays the sound on your computer. You must have a sound player
application installed to hear the sound. For sounds in the gsm format, Quicktime by
Apple is a good player.
Play Sound To Phone Icon. Displays the Find an Extension popup menu. Search
for and enter the extension to ring, then click Go. This rings the extension, then plays
the message when you pick up the phone.
Modify Icon. Displays the Sound Information details. Make your changes, then
click Save Sound. The Sound Manager main page is displayed with a message
indicating that the sound was modified. Supply or change the following fields:
• Sound Name. Change name if desired.
• Sound Description/Script. What are the exact words of this recording? Change
the script or description.
• Folder. Cannot change here.
• Sound Language. Cannot change here.
• Sound Source. Select from the dropdown menu either Record Over Phone or
Upload File.
• Upload File. If this was selected, click to display files on your hard drive.
• Extension to Ring for Recording. Search for and enter the extension if you
selected Record Over Phone. Then click Ring Phone and make your recording,
then hang up the phone or press #..
WARNING: If you are uploading or re-recording the sound file, the existing file
is replaced. You will not be able to retrieve the old version of the file.
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Media
Deleting a Sound
In the sounds list on the main page, check the boxes for the sounds that you want to
delete, then click the Delete icon (at the top of the song list). Be sure that you do
indeed want to delete these sounds, then click Yes, Delete Sounds. The main page is
displayed again, with a message that the sound has been deleted.
NOTE: If a sound box cannot be checked, it is because it cannot be deleted. The
basic set of system sounds cannot be deleted.
Moving a Sound into
a Folder
In the sounds list, check the boxes that you want to move, select the folder the sound
belongs in (the dropdown menu), then click the Move Checked To icon. The Sound
Manager main page is displayed with a message that the sound has been moved. If
you have used the sound in IVRs or call queues, Switchvox now accesses the sound
from its new folder.
NOTE: If a sound box cannot be checked, it is because it cannot be moved. The
basic set of system sounds cannot be moved.
About Sound Packs
Sound packs are pre-configured packages that you can add to Switchvo and use in
place of the pre-installed default English sound pack. Only officially certified sound
packs can be installed on Switchvox. To obtain sound packs for different languages,
go to Switchvox Product Addons.
To add a sound pack:
•
•
•
•
•
•
Download the sound pack to your desktop computer.
Go to Tools > Media > Sound Manager.
Select Sound Pack from the Create dropdown.
Click Go. This displays the Sound Pack Information page.
Click the text box to locate the file on your computer.
Click Upload Sound Pack.
The sound pack is added to the Sound Manager, and the language of the sound pack
is available for use.
Managing Folders
Sounds are organized in folders. Switchvox comes preconfigured with several folders,
which cannot be deleted. Typically, the folder name indicates the type of sounds it
contains. Any new folders should be named accordingly.
Adding a New Folder
To create a new folder, select Folder from the Create dropdown menu, then click Go.
Enter a unique Folder Name and a Description for this folder, then click Save Sound
Folder. This displays the Sound Manager main page with a message that a new folder
has been created, and the new folder is included in the folders list.
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Modifying a Folder
To modify a folder, click its Modify icon. Make your changes, then click Save Sound
Folder. The Sound Manager main page is displayed with a message that the folder has
been modified.
Deleting a Folder
To delete a folder, select its Delete icon. Be sure that you do indeed want to delete this
folder and all of the sounds in it, then click Yes, Delete Folder. The folder and all
sounds in it are permanently removed, and the Sound Manager page is displayed with
a message that the folder has been deleted.
Managing Languages
Switchvox supports multiple languages. Each sound in the Sound Manager can be
uploaded or recorded in all the available languages. Your sounds ‘hi,’ ‘hola,’ and
‘ciao’ sounds are grouped together, because they say the same thing to the caller.
The pre-configured default language is English, and there is a set of system files
tagged English.
Adding a New Language
To add a new language, select Language from the Create dropdown menu. Click Go.
This displays the Language Information page. Enter the Language Name, then click
Save Language. The Sound Manager main page is redisplayed with a message that a
new language has been added.
Modifying a Language
To modify a language, click its Modify button. Make your changes, then click Save
Language. The Sound Manager main page is displayed, with a message that the
language was modified.
Deleting a Language
To delete a language, click its Delete icon. Be sure that you do indeed want to delete
this language and all of the sounds associated with it, then click Yes, Delete
Language. The Sound Manager main page is displayed with a message that the
language was deleted.
Setting a Default Language
To set a language as the default language, click its Default Language check box. The
page is refreshed with a message that the sound pack was successfully activated. Now
the sounds labeled with that language are played. If there is no sound available in that
language, Switchvox uses the English system sound-file.
WARNING: Changing the Default Language automatically causes the Switchvox software to restart. All current phone connections will be dropped, and members will be logged out of all call queues.
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Reporting
Reporting is the tool that generates detailed usage reports about your calling patterns
and queue activity. You can design custom reports for special reporting requirements.
Call Reports
103
Queue Reports
Custom Reports
106
108
Call Reports
Concurrent Calls 103
Call Logs 104
Call Reports 105
Concurrent Calls
The Concurrent Calls Report displays the total calls happening at the same time,
over a user-specified date and time range.
Report Criteria
Specify the reporting period by indicating the following:
• From Date and To Date
• From Time and To Time
Report Types
Then select one of the following report types and indicate the setting on which to
report:
• By Call Type. After selecting this type, indicate one of the following settings on
which to report:
-All Calls reports for all calls.
-Internal reports for only calls made within the organization.
-External reports for only calls made to the outside.
• By Origination. After selecting this type, indicate the setting on which to report.
-Origination. Select either Outgoing or Incoming calls.
-Provider. Select All or select from the list of providers.
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Call Reports
• By Provider. After selecting this type, either indicate All or select a provider from
the list.
When finished, click Chart Report to generate the results in chart form.
Call Logs
The Call Logs Report lets you generate a log of call activity for a date-range and a
user-specified type. Results can be viewed in your browser or output to either an.xls
file or XML file.
Log Criteria
Specify the reporting period by indicating the following:
• From Date and To Date
Log Type
Then select one of the following log types and set the entity to log:
• All Calls. Reports all calls in your system between the dates specified. Select a
different type for more specific logs.
• Phones. Select from the list of extensions, the extensions for which you wish to
log call activity.
• Call Queues. Select from the list of queues, the queues for which you wish to log
call activity.
• IVRs. Select from the list of IVRs, the IVRs for which you wish to log call activity.
• Providers. Select from the list of providers, the providers for which you wish to
log call activity.
• Caller ID Settings. Enter either a Caller ID Name or a Caller ID Number for
the caller for which you wish to log call activity..
NOTE: Keep your date range small because Caller ID searches take extra processing power over a large number of calls.
• Incoming DID. Enter an incoming DID.
NOTE: Keep your date range small because DID searches take extra processing
power over a large number of calls.
Output Options
When finished, click one of the following to generate the log:
• View Log opens the report in the browser window.
• Output to .xls file downloads an .xls file that you can open with an application
such as Microsoft Excel.
• Output to XML downloads an XML file.
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Call Reports
Call Reports
Call Reports lets you generate a report about call activity for the date-range,
breakdown, and type you specify.
Report Criteria
From Date and To Date. Enter the date range in mm/dd/yyyy format.
Ignore Weekends. Select YES if your organization is not active on weekends and
calls on those days can be ignored.
Report Breakdown. Indicate from the dropdown menu how you wish the report to be
broken down:
•
•
•
•
•
Report Type
By Date
By Account
By Hour of Day
Day of Week
Total Cumulative
You can select what fields of data you want to include in a report, depending on what
calls you include in the report. Select one of the following types, then set the desired
entity to report on:
• Overview. Show all calls in your system between the dates indicated (in Report
Criteria):
-Total Number of Calls.
-Total Number of Incoming Calls
-Total Number of Outgoing Calls
-Total Talking Time. Talk time means the time that the callers were actually
talking to each other.
-Total Call Duration. Call duration means the entire call, including time spent
in an IVR or waiting in a queue.
-Average Talk Time per Call
-Average Call Time per Call
• Phones. Select from the list of extensions, the extensions for which you wish to
report call activity. The same subcategories as listed for Overview are available.
• Call Queues. Select from the list of queues, the ones for which you wish to report
call activity.
• IVRs. Select from the list of IVRs, the IVRs for which you wish to report call
activity.
• Providers. Select from the list of providers, the providers for which you wish to
report. You can create reports for the same subcategories as listed for Overview
• Incoming DID. Enter an incoming DID. You can create reports for the same subcategories as listed for Overview.
NOTE: For DID settings, keep your date range small because DID searches take
extra processing power over a large number of calls. Also, the fewer digits you
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enter, the more processing power will be required to perform the search, so more
digits is better.
Report Output
Options
To generate the report, you can choose one of the following output options.
Chart Report. Open the report as a graphical chart in the browser window.
View Report. Open the report in the browser window.
Output to .xls File. Download a .xls file that you can open with an application such
as Microsoft Excel.
Output to XML. Download an XML file.
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Queue Reports
Queue Reports
Queue Logs 106
Queue Reports 107
Queue Status 107
Queue Logs
Queue Logs return a list of queue calls based on the criteria and type you specify.
Log Criteria
From Date and To Date. Enter the date range in mm/dd/yyyy format.
Ignore Weekends. Select YES if your organization is not active on weekends and
calls on those days can be ignored.
The log will include queue calls happening during this date range.
Queue Type
Queues. Select this type to include calls made to a selected queue.
Queus Members. Select this type to include calls made to selected queue members.
To select multiple members, hold down the CTRL key and click on the desired
members.
Queue Reports
Queue Reports return data according to the criteria and type you specify. When
finished, click the report type button.
Report Criteria
Enter the From Date and To Date in the mm/dd/yyyy.
Ignore Weekends. Select NO or YES.
The report will include queue calls happening during this date range.
Queue Type
Queues. Select this type to include calls made to the selected queue.
Queue Members. Select this type to include calls made to the selected queue members. To select multiple members, hold down the CTRL key and click on the members.
Queue Report Output
Options
To generate the report, you can select one of the following output options.
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Chart Report. Open the report as a graphical chart in the browser window.
View Report. Open the report in the browser window.
Output to XML. Download an XML file.
Queue Status
Queue Status shows the current status of a queue, and today's statistics for the queue
and its members. For a real-time view of a queue, use its Switchboard Queue Panel.
For historical information about a queue, see Queue Reports and Queue Logs.
Queue Status provides real-time information for each call queue in Switchvox:
• Overall statistics.
• Member status and statistics.
• Calls waiting in the queue.
The information in Queue Status is only for one day. For historical information, see
Queue Logs or Queue Reports.
Statistics
Queue Statistics show you how a queue has been functioning for the following categories:
• Current Calls Waiting. Calls waiting in the queue.
• Completed Calls. Calls answered by a queue member.
• Abandoned Calls. Calls in the queue, but hung up before being answered by a
queue member.
• Redirected Calls. Call was redirected out of the call queue, or the caller pressed 0
key, or the call reached maximum timeout, or the call was sent the maximum number of times, or the call queue was at its maximum length.
• Longest Queue Length. Largest number of calls waiting in the queue at one time.
• Average Entry Position. Average position when calls first enter the queue.
• Longest Wait Time for Completed Calls. Longest time that a call was in the
queue and was answered by a queue member.
• Longest Wait Time for Abandoned Calls. Longest time that a call was in the
queue but hung up before being answered by a queue member.
• Average Wait Time for Completed Calls. Average time that a call was in the
queue and was answered by a queue member.
• Average Wait Time for Abandoned Calls. Average time that a call was in the
queue but hung up before being answered by a queue member.
• Average Talk Time. Average time that queue members spent talking with callers
from the queue.
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Member Status
Member Status shows each member of the queue, their extension and queue-status
(logged in, logged out, or logged in but paused), who they are talking to, and the
amount of time they spent on their last call. It also includes statistics: the number of
calls they have taken today, and the average time they spent talking on those calls.
Current Calls Waiting
Current Calls Waiting shows the calls that are currently waiting in the queue,
including the current position, caller ID, how long they have been waiting in the
queue, and the position in which they entered the queue.
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Custom Reports
Custom Reports
Scheduled Reports
Scheduled Reports is similar to Call Reporting, but with this tool you can set up a
report to run at a certain time with a particular output, and the report is sent to you (via
email) based on those criteria. To edit an existing scheduled report, click Modify on
the same line as the report.
Create Scheduled Call
Report
Click Create Scheduled Call Reports button to create a new call report. This displays the Create Scheduled Call Report page, which has three tabs: General Settings, Schedule Information, and Report Parameters.
General Settings tab
Report Name. Enter the report’s name.
Output Type. Select the output type from the dropdown menu: Chart, HTML, or
XML file.
Delivery Email Where the report will be sent. Enter the full email address where
you wish to send the report.
Schedule Information tab
Schedule Time Frame. Enter the Start Date and End Date that you want to activate
the report (the first date it should run) and the date that you want to deactivate it (the
last date it should run). If you want the report to keep running indefinitely, select YES
for Run Forever.
Schedule Occurrence. The occurrence pattern indicates how often the report should
run during the time it is active. Click the desired option button:
•
•
•
•
•
Hourly. Enter the desired Hourly Settings.
Daily. Enter the desired Daily Settings.
Weekly. Enter the desired Weekly Settings.
Monthly. Enter the desired Monthly Settings.
Yearly. Enter the desired Yearly Settings..
Report Parameters tab
Report Parameters . Select NO or YES to Ignore Weekends in report and select
Breakdown report by one of the following: Date, Account, Hour of Day, Day of
Week, or Total Cumulative.
Report Type. Select one of the following types:
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Custom Reports
• Overview settings show all calls in your system between the dates you have set.
Choose a different type for more specific reports.
• Phones settings list the extensions and names in your system and will report calls
for these individuals in your system between the dates you have set.
• Queues settings list the queue extensions in your system and will report calls for
each queue in your system between the dates you have set.
• IVRs settings list the IVR extensions in your system and will report the total number of calls for each between the dates you have set.
• Providers settings list the SIP providers and their names and will report the calls
for each provider in your system between the dates you have set.
Click Save Report Settings when finished.
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Server
This section describes the tools under the Server category, which describe how to perform maintenance tasks, manage your server's network settings, and diagnose issues.
Diagnostics 106
Maintenance 113
Networking 117
Diagnostics
The Diagnostics tools give you views of the call activity in Switchvox and help you
evaluate potential problems:
Connection Status 106
Current Calls 108
Server Information 108
Disk Usage 109
SNMP 109
Error Logs 112
Advanced Debugging 112
Technical Support 112
Connection Status
This tool displays Connection Status information for the following entities connected
to Switchvox:
•
•
•
•
•
VOIP Providers
VOIP Providers
SIP Phones
Hardware Devices
Peered Switchvoxes
Fax Print Jobs
This tool lists the following information for each VOIP provider:
• Type
• Name of the provider as entered when the provider was created
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Diagnostics
•
•
•
•
•
Host that Switchvox connects to when registering and sending calls
Account ID
Callback Extension that incoming calls are sent to by default
Latency (ms)
State of the connection. This value is Registered when the system is operating
properly. If it is not, outgoing calls will fail and incoming calls cannot be received.
Here are some common reasons a VOIP provider is in a non-registered state and a
resolution:
-VOIP provider has just been entered and the registration has not finished yet.
Wait a few minutes and then refresh the page.
-Username or password are incorrect. Check to make sure you have entered
them correctly.
-Firewall is not allowing the communication between Switchvox and your provider to flow freely. When you use a SIP-aware firewall, it recognizes the data
being sent back and forth. A normal router does not require any special configuration, but a SIP-aware firewall does. Contact the manufacturer of the firewall
for details on setting up any necessary routing rules.
SIP Phones
This tool lists all phone Extensions with their respective Caller ID, User Agent, IP
Address, and Phone State.
Phone States. The following are possible phone states:
• Registered. This indicates the phone is configured properly and is on the network.
• Timeout: This indicates the phone was registered at some point, but the server no
longer recognizes it and does not send calls to this phone anymore.
• Unreachable. This indicates the phone is not registered with Switchvox.
If a phone is not registered but it should be, check the following:
-The phone should be set to register every 120 seconds. Some phones are by
default set to only register once. See your phone’s user manual for more information about this setting.
-The phone must know the Switchvox IP address, and it must have the same
password as the "Phone Password" for the extension. For more information on
configuring your phone, see your phone's manual or call your phone's technical
support.
-The phone must be ON, its Ethernet cord plugged into the phone and the correct
Ethernet jack, and it must be on the same LAN as Switchvox (not on a separate
partitioned LAN subnet).
-There should be no firewall between your phone and Switchvox that could
block your phone from accessing Switchvox.
NOTE: If a SIP phone is not in a Registered state, click its Diagnose button to
run a diagnostic tool for the phone.
External Phones. Phones are considered External if they are not on the same network
as Switchvox. This generally means they have to connect to Switchvox over the internet, or they are located on a separately partitioned network. External phones cannot
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Diagnostics
go through the full diagnostics because Switchvox does not have direct access to these
phones.
Finding the IP Address of your SIP Phone. To run the diagnostic tool, you need to
know the IP address for your phone.
-Polycom Soundpoint IP 300/301/500/501/600/601 Each phone is set up to
obtain its IP address via DHCP. You can find the IP address of a Polycom phone
by pressing the Menu button on your phone. Then, on the LCD screen of your
phone highlight “Status...” and press select and then highlight “Network...” and
press select again, then highlight “TCP/IP Parameters...”and press select. Now
find the line that starts with “IP:” and write down the IP Address.
-Sipura ATA. Connect a phone to the jack labeled “Phone 1” on the front of your
ATA. Then on the phone dial “****” and you will then hear a voice say “Configuration Menu”. Then dial “110#” and it will read back the IP address of your
ATA.
Hardware Devices
This tool lists all the channels on the hardware devices installed for Switchvox and an
alarm state for each. The state of these channels, if properly connected, should be No
Alarm.
If any other alarm state is shown (a RED Alarm or a Loss of Signal), check that the
device is plugged into the correct line and that you have the appropriate signaling type
selected.
Peered Switchvoxes
This tool lists any peered Switchvoxes with each peer’s Name, Host, Status Details,
and available Actions. This tool is typically used after creating peered Switchvoxes
during setup. If Status Details reports any errors, click the Diagnose icon to execute a
diagnosis of the Peered Switchvox. This lists the tests performed and the results
(Passed, Failed, Not Tested, and Warning) followed by a comment. You can Return to
the Connection Status page, make the correction and repeat the Diagnose action.
Fax Print Jobs
This tool lists the current Print Jobs for faxes. These Print Jobs occur when an extension-owner is creating a fax from his or her desktop computer.
Current Calls
This tool lists all current calls. It shows the person who called, the person who took
the call, the duration of the call (in seconds), the current state of the call, and the provider. You can select the time interval for refreshing the call list. The default is to
refresh the list every 60 seconds.
To hang up on a call, click Hangup, then click Yes, Hangup.
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Diagnostics
Server Information
This tool displays information about the Switchvox server. As long as the Current Status for each item is Good, the hardware is fine.
If a field is over threshold, it needs immediate attention.
memory. If memory is over threshold, one option is to turn the system off (wait a
few moments), then turn it back on. This normally frees up the memory usage.
Another option is to install more RAM. You can also attempt to identify the process
that is overloading the memory (call recording, concurrent calls, concurrent conference calls, etc.), and stop that process.
disk. If disk is over threshold, check voicemail, fax, call recordings, and backups to
find out what is taking up the disk space.
load. If load is over threshold, check Diagnostics > Current Calls to see if the system is hitting a peak (spike) amount of concurrent calls.
Advanced Server Information. A Switchvox Technical Support representative may
ask you check this area for a specific issue. In most cases, you will not need to use this
tool.
Disk Usage
Disk Usage has two tabs: Disk Usage and Mailbox Usage.
Disk Usage
Breakdown tab
Disk Usage Breakdown gives the percentage that the Disk is full and displays a
color-coded bar graph showing the amount of disk space taken by the following:
•
•
•
•
•
•
PBX Software
Config/Logs
Media
Voicemail/Fax
Call Recordings
Backups
For information about disk-space quotas, see General Settings (Setup > Extensions >
Manage > Phone and Virtual Extensions > Profile Information > Advanced
Options > General Settings).
Mailbox Usage
Breakdown Tab
Mailbox Usage Breakdown displays Voicemail and Fax usage by extension, including Name, Quota Class, Quota Used, # Of VM, VM Usages, # Of Faxes, and Fax
Usage. You can search for an extension by extension number or name.
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Diagnostics
SNMP
SNMP (Simple Network Management Protocol) is used to monitor devices on an IP
network. If you are not already using SNMP, then you can ignore these settings.
If you are already using SNMP, you can monitor the OIDs that the Switchvox agent
provides.
IMPORTANT: Be sure to allow SNMP traffic in Switchvox Access Control. (By
default, SNMP traffic is not allowed).
SNMP v1/v2c Community Name. Enter the Community Name for the SNMP community to which Switchvox should belong.
SNMP v3 User Name. Enter the SNMP User Name if you are using SNMP
version 3.
SNMP v3 Password. Enter the SNMP Password if you are using SNMP version 3.
Switchvox OIDs
The Switchvox prefix is .1.3.6.1.4.1.22736.10
.1 Switchvox Information
.1.3.6.1.4.1.22736.10.1.1
software version
.1.3.6.1.4.1.22736.10.1.2
max concurrent calls
.1.3.6.1.4.1.22736.10.1.3
used feature packs
.1.3.6.1.4.1.22736.10.1.4
max feature packs
.1.3.6.1.4.1.22736.10.1.5
used extensions
.1.3.6.1.4.1.22736.10.1.6
max extensions
.2 VOIP Providers
.1.3.6.1.4.1.22736.10.2.1
number of VOIP providers
.1.3.6.1.4.1.22736.10.2.2
number of state 'Ok' VOIP
providers
.1.3.6.1.4.1.22736.10.2.3.1.1
first VOIP provider id
.1.3.6.1.4.1.22736.10.2.3.1.2
first VOIP provider type
.1.3.6.1.4.1.22736.10.2.3.1.3
first VOIP provider name
.1.3.6.1.4.1.22736.10.2.3.1.4
first VOIP provider host
.1.3.6.1.4.1.22736.10.2.3.1.5
first VOIP provider
account id
.1.3.6.1.4.1.22736.10.2.3.1.6
first VOIP provider callback extension
.1.3.6.1.4.1.22736.10.2.3.1.7
first VOIP provider latency
.1.3.6.1.4.1.22736.10.2.3.1.8
first VOIP provider state
.1.3.6.1.4.1.22736.10.2.3.[2-xxx].[1-8]
details for each additional
VOIP provider
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.3 Hardware Status
.1.3.6.1.4.1.22736.10.3.1.1
memory used
.1.3.6.1.4.1.22736.10.3.1.2
memory total
.1.3.6.1.4.1.22736.10.3.1.3
swap used
.1.3.6.1.4.1.22736.10.3.1.4
swap total
.1.3.6.1.4.1.22736.10.3.1.5
last update (memory/swap)
.1.3.6.1.4.1.22736.10.3.2.1
disk used
.1.3.6.1.4.1.22736.10.3.2.2
disk total
.1.3.6.1.4.1.22736.10.3.2.3
last update (disk)
.1.3.6.1.4.1.22736.10.3.3.1
load
..4 Phone Status
.1.3.6.1.4.1.22736.10.4.1
number of SIP phones
.1.3.6.1.4.1.22736.10.4.2
number of reachable SIP
phones
.1.3.6.1.4.1.22736.10.4.3.1.1
first phone id
.1.3.6.1.4.1.22736.10.4.3.1.2
first phone extension
.1.3.6.1.4.1.22736.10.4.3.1.3
first phone caller id
.1.3.6.1.4.1.22736.10.4.3.1.4
first phone user agent
.1.3.6.1.4.1.22736.10.4.3.1.5
first phone ip address
.1.3.6.1.4.1.22736.10.4.3.1.6
first phone state
.1.3.6.1.4.1.22736.10.4.3.[2-xxx].[16]
details for each additional
phone
.1.3.6.1.4.1.22736.10.4.2
number of reachable SIP
phones
.5 Current Calls
.1.3.6.1.4.1.22736.10.5.1
total current calls
.1.3.6.1.4.1.22736.10.5.2
number of calls in state
'agent_login'
.1.3.6.1.4.1.22736.10.5.3
number of calls in state
'agent_logout'
.1.3.6.1.4.1.22736.10.5.4
number of calls in state
'checking_voicemail'
.1.3.6.1.4.1.22736.10.5.5
number of calls in state 'conference'
.1.3.6.1.4.1.22736.10.5.6
number of calls in state 'directory'
.1.3.6.1.4.1.22736.10.5.7
number of calls in state 'faxing'
.1.3.6.1.4.1.22736.10.5.8
number of calls in state 'intercom'
.1.3.6.1.4.1.22736.10.5.9
number of calls in state 'ivr'
.1.3.6.1.4.1.22736.10.5.10
number of calls in state
'leaving_voicemail'
.1.3.6.1.4.1.22736.10.5.11
number of calls in state 'monitoring'
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.1.3.6.1.4.1.22736.10.5.12
number of calls in state 'parked'
.1.3.6.1.4.1.22736.10.5.13
number of calls in state 'queued'
.1.3.6.1.4.1.22736.10.5.14
number of calls in state
'receiving_fax'
.1.3.6.1.4.1.22736.10.5.15
number of calls in state 'ringing'
.1.3.6.1.4.1.22736.10.5.16
number of calls in state 'talking'
.1.3.6.1.4.1.22736.10.5.17
number of calls in state 'unknown'
.6 Telephony Cards
.1.3.6.1.4.1.22736.10.6.1
number of devices
.1.3.6.1.4.1.22736.10.6.2
number of state 'Ok' devices
.1.3.6.1.4.1.22736.10.6.3.1.1
first device id
.1.3.6.1.4.1.22736.10.6.3.1.2
first device name
.1.3.6.1.4.1.22736.10.6.3.1.3
first device channel
.1.3.6.1.4.1.22736.10.6.3.1.4
first device channel group
.1.3.6.1.4.1.22736.10.6.3.1.5
first device signalling
.1.3.6.1.4.1.22736.10.6.3.1.6
first device state
.1.3.6.1.4.1.22736.10.6.3.[2-xxx].[1-6]
details for each additional device
.7 Switchvox Subscription
.1.3.6.1.4.1.22736.10.7.1
expire date
.1.3.6.1.4.1.22736.10.7.2
days left
.8 RAID Controller
Other OIDs
1.3.6.1.4.1.22736.10.8.1
controller status
1.3.6.1.4.1.22736.10.8.2
logical device status
1.3.6.1.4.1.22736.10.8.3
device #0 status
1.3.6.1.4.1.22736.10.8.4
device #1 status
The Switchvox agent provides some additional OIDs that are not specific to the
Switchvox application.
.iso.org.dod.internet.mgmt.mib-2.system
Linux system OIDs. You can use any OID under this.
.iso.org.dod.internet.private.enterprises.ucdavis
Linux system OIDs. You can use any OID under this.
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Diagnostics
Error Logs
These tools are for our customer service representatives to help you with problems
you might have with Switchvox. You do not need to use these tools unless we ask you
to.
The tools included here are simple error logs, advanced error logs, and advanced
debugging sessions.
Advanced Debugging
We recommend that you do not use this tool unless you are being directed by a technical support representative.
These debugging sessions collect detailed PCAP, PRI, or ACLI information to help
you see exactly what is happening in Switchvox.
The information can be useful in identifying the source of a problem; however, these
sessions place an additional load on Switchvox system resources.
Technical Support
These tools are for Switchvox Technical Support representatives to help you with
problems that you might have with Switchvox.
IMPORTANT: You do not need to, and should not use these tools unless it is at
the request of a Switchvox Technical Support representative.
Technical Support Access lets us access your machine to help diagnose a problem.
You must enable Technical Support Access to use the VPN connection. If you disable
it, then all access to your system will be closed, including an open VPN connection.
A Technical Service Request (TSR) can be downloaded, then sent to Switchvox Technical Support.
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Maintenance
Maintenance
The following system Maintenance tools are available:
Telephony Cards 113
Backups 114
Registration 115
Updates 115
System Clock 117
System Reload 117
Telephony Cards
The Telephony Cards tools let you scan for and configure new telephony hardware in
the Switchvox server. You can also delete hardware. If hardware has not been
detected, make sure all cards are installed properly and that the card you are using is
supported. See the Digium web site for installation instructions.
Hardware Devices. Each device connected to the Switchvox server is listed under
this section. To configure a listed device, click Configure for that device. To delete a
listed device, click Delete for that device, then verify the deletion request. Devices are
listed with Card Type, Device Type, Device Name, Spans (the logical groupings of
channels), and Total Channels.
If you have added new hardware, Switchvox detects it when you start the server. If
you have deleted a device and want Switchvox to detect it again, click Scan for New
Devices. Switchvox detects the telephony hardware and includes it in the Hardware
Devices list.
Hardware Country Settings. If you need to change the country in which your
Switchvox server is located, select it from the dropdown, then click Save Country.
This setting also sets the call-progress tone scheme that determines the tones sent to
your callers.
Hardware Echo Cancellation. This option disables the hardware’s echo cancellation
so all echo cancellation is handled by Switchvox. For details about those settings, see
“Advanced Options” page 41 under Channel Groups in Call Routing.
Echo Reduction for Analog Lines. This tool allows Switchvox to run a series of
tests, after which it calibrates the analog lines to the results. Click Run Echo Reduction Tool to run. We recommend running Echo Reduction if you move lines to other
ports, or add new lines.
NOTE: Make sure all your analog lines are plugged into your PBX and have a
dialtone before running the echo reduction tool.
WARNING: Running this tool causes your system to go offline for approximately two minutes per analog line. Any current phone calls are dropped, and any
agents logged in to call queues are logged out.
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Maintenance
Backups
The Backups tool lets you create and download a backup file containing much of the
data and audio files for Switchvox. This file may be used at any time to restore
Switchvox to the same state it was in at the time the backup was created.
NOTE: Always backup your Switchbox system before making any changes to it,
including restoring the system from a backup.
IMPORTANT: A best practice for backups is to schedule a daily automatic
backup that stores the backup file on an FTP server. For details, see “Set Up
Automatic Backups” page 122.
Create Backup
Create Backup displays the Create Backup File page. Select any optional components you want to include in the backup.
NOTE: Call Recordings are backed up separately. For details, see “Call Recording” page 81.
When ready to begin, click Save Backup.
WARNING: The backup process may take several minutes depending on the
amount of content. Do not click any links or buttons during the process.
After the backup is finished, the Backups page is displayed with the new backup listed
in the Existing Backups table. Click Download to save the Backup (which may
include multiple files). Store the Backup in a location from which it can be easily
accessed if necessary to restore your system.
Set Up Automatic
Backups
This tool displays the Set Up Automatic Backups page. Enter the following information:
Schedule Settings
• Automatic backups will occur every. Select the day of the week and enter the
time to run the backup process.
• Number of backups to keep. Enter the number of backups to store. When this
number is exceeded, the oldest backup is deleted from the FTP host.
FTP Settings
Enter settings for the FTP server to store backups:
• FTP Host Name. Enter the hostname of your FTP server.
• FTP Username and FTP Password. Enter the username and password to use to
access the server
• FTP Directory Path. Enter the path to the directory to store the backup files.
Click.
• Use SFTP. Select NO or YES.
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• Test FTP Settings. Click to test this information to be sure your backup files can
be transferred.
After you have entered the backup information, click Save Settings.
To disable automatic backups, clear the FTP information, then click Save Settings.
Upload and Restore
Backup
Be absolutely sure you want to restore your system from any backup file before clicking a Restore option because restoring your system from a backup file causes the following to happen:
-Your current Switchvox configuration is overwritten with the contents of the
selected backup file.
-All current calls are dropped.
-All agents will be logged out of queues.
-All data created since the backup was created will be lost, including extensions
and voicemail.
• Restoring From the List.
If the backup file you want to use is listed in the table of Backups, you can restore
from the list. Find the backup file in the list and click its Restore icon. Verify you
want to restore using this backup, by clicking Yes, Restore Backup.
• Restoring From a Backup on Your Computer.
If the backup file you want to use is not listed, then you need to upload the backup
file from your hard drive, then restore. To use this process, do the following:
-Browse to find the file on your computer. If the backup is contained on multiple
files, select the meta file.
-If you select a meta file, you are prompted to Browse for all of the files.
-After selecting the files, click Upload and Restore Backup.
• Deleting a Backup File from the List.
-Click the Delete button for the backup.
-Verify you want to permanently delete the file by clicking Yes, Delete Backup.
This deletes and removes the backup from the list, and the Backup page is displayed with a message indicating that the file has been deleted.
Registration
Registration is usually completed when Switchvox is first installed. If this is the case,
this page displays the first few characters of your Registration Code, which you will
need if you call Switchvox Technical Support.
If you have not yet registered Switchvox, please do so.
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Maintenance
Updates
This tool displays the following:
Current System Info
• Current Version. Version of Switchvox software installed with a link to the
Release Notes.
• Maximum User Extensions. Total number of extensions and number currently
being used.
• Phone Feature Packs. Total number of feature packs and number currently being
used. Phone Feature Packs let you easily configure your phones for use with
Switchvox, and offer great tools for customizing your phones. You need one
Phone Feature Pack per SIP phone. See your reseller for instructions on purchasing and activating more Phone Feature Packs. If you purchased your phones with
Switchvox, then each phone already has its own Phone Feature Pack and is already
configured for use with Switchvox. Phone Feature Packs are not necessary for any
other type of extension.
A Switchvox Phone Feature Pack offers the following features:
-Customized phone displays. Customize some of the information displayed on
your phone. See “Phone Features” page 13 in the Switchvox Extension-Owner
User Guide.
-Status Indicators. Set a line status indicator light on your phone. This can
show whether a call is waiting in a parking lot, or whether you are logged into a
queue. See “Status Indicator” in the Switchvox Extension-Owner User Guide.
-Distinctive Ringtones. Set the audio file of your choice to ring on incoming
calls, depending on the caller. See “Ring Rules Tab” page 18 in the Switchvox
Extension-Owner User Guide.
-Auto-answer. Set an automatic answer in Intercom mode, depending on your
caller. See “Ring Rules Tab” page 18 in the Switchvox Extension-Owner User
Guide.
-Combine Ringtones and Auto-answer. Combine a ringtone and auto-answer,
depending on the caller. See “Ring Rules Tab” page 18 in the Switchvox
Extension-Owner User Guide..
-Phone Applications. Access many of the Switchvox features right from your
phone.
The Phone Features available on your phone vary, depending on the type of
phone you have.
-General Options are available on snom and Polycom phones.
-Other features are only available on Polycom phones. If you have a newer Polycom phone with three or more lines (e.g., the VVX 1500 or SoundPoint 450,
550, 560,650, or 670) you should be able to take full advantage of your Phone
Feature Pack. Other Polycom phones may not offer all of the features.
-If you use Switchvox remotely (your phone is not on the same network as
Switchvox), you will not see Switchvox Caller Profiles.
• Subscription Expiration. If your Switchvox subscription is about to expire, then
you will see alerts and options to renew your subscription. If you renew your subscription, those alerts go away. Or, you can turn the alerts off at any time.
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Available Updates
Updates. If no Updates are displayed, then you already have the newest version of the
software.
If Updates are displayed, each update has two actions: Download, and Download and
Apply.
-Download. This option downloads the update to Switchvox with a progress
indicator so that you can see how long it will take. After the download is complete, the update is ready to be applied, and the button Apply This Update
appears, You can apply the update whenever it is convenient. After you apply
an update, the files for that update, and any previous updates that you may have
downloaded, are removed from Switchvox.
-Download and Apply. If you click Download and Apply, the update is downloaded to Switchvox and immediately applied.
Obsolete Updates. Lists obsolete updates.
System Clock
System Clock shows you the current time zone and time setting. By default, Switchvox uses NTP to retrieve its time settings. Unless you have a particular reason to
change this, leave your clock set to the default NTP settings. The time zone can be
changed if the machine is moved from one location to another.
System Reload
We recommend that you do not use these options unless instructed to do so by a Technical Support representative.
If you decide to use these tools to reload, restart, reboot, or shut down Switchvox, first
make a system backup file (see “Create Backup” page 122), then inform your
Switchvox users that calls will be dropped and members will be logged out of their
queues.
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Networking
Networking
Networking includes tools to manage the following:
IP Configuration 117
HTTP and Proxy 117
Access Control 120
Phone Networks 123
IP Configuration
This tool lets you set the IP configuration for Switchvox. These settings are similar to
settings you would set on any other computer on your network. Enter the following
information, then click Save IP Configuration.
Gateway Address. Enter the IP address of the machine to which Switchvox sends
outbound traffic. It is typically the address of your router.
DNS Addresses. Enter up to three DNS addresses. Switchvox uses these addresses to
translate any host/domain names into IP addresses.
Allow NAT Port Forwarding. Select YES to allow NAT Port Forwarding to Switchvox. This option is useful if you need to handle calls going to and coming from an
external network, and Switchvox is behind a router that performs NAT.
On your router, you must forward the following ports to your Switchvox server. Also,
in Machine Admin > Access Control, you must have a rule that allows the appropriate
traffic for an external network. See “Access Control” page 128 for more information.
Protocol
TCP/
UDP
Port(s)
Description
IAX
UDP
4569
Signaling port needed for phones outside of your network.
SIP
UDP
5060
Signaling port needed for phones outside of your network.
SIP
UDP
10000-10500
RTP audio ports needed for phones
outside of your network.
TCP
80
HTTP port for remote web voicemail
access
TCP
443
HTTPS port for remote web admin
access
TCP
5222 & 843
Ports for using the Switchboard
remotely
TCP
5269
Port for remote XMPP (Jabber/chat)
access
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Networking
External IP Address. Enter the public IP Address of your router. If you are not sure
what your public IP is, click Look Up External IP to automatically find it.
NOTE: SIP phones that are outside of your network must use this external IP for
registration.
You cannot use this feature if your ISP does not give you a static external IP address.
Check with your ISP to make sure that the public IP you enter here will not change. If
you do not have a static public IP, you will only be able to use phones from within
your local network.
eth0 Interface. Enter the IP and Netmask address. You set these addresses when you
installed the Switchvox software, and you do not normally need to change this information after installation.
Advanced Options. If you need to change an Advanced Option, click Show
Advanced Options and enter the following information:
• Hostname. You can set a hostname for Switchvox (e.g., pbx.example.com) if you
have a DNS setting. This may alleviate problems with delivering emails through
particularly strict SMTP servers.
IMPORTANT: This option also sets the web server’s SSL key. This may help prevent warnings from your browser when logging into Switchvox. In most cases it is
best to leave this option unchanged.
• Jabber Hostname. You can set a hostname for the Jabber server in Switchvox
(e.g., jabber.example.com) if you have a DNS setting. This is important if you are
peering Switchvoxes and want to use functions such as Presence or the Chat Panel.
• IP ToS Audio. This lets you set the ToS or DSCP field in VoIP packets sent by
Switchvox. This field can be used by firewalls and switches to distinguish specific
types of traffic to apply QoS rules, such as favoring all voice traffic for better quality. Setting the ToS field isn’t a requirement for prioritizing VOIP traffic on your
router, it’s just one way to identify VOIP traffic. If you’re not explicitly prioritizing this field in your network equipment, changing this option will have no effect
on your VOIP quality.
For more information see the following resources:
-RFC 2474. Definition of the Differentiated Services Field (DS field) in the IPv4
and IPv6 Headers, Nichols, K., et al., December 1998.
-IANA Assignments, DSCP registry, Differentiated Services Field Codepoints:
http://www.iana.org/assignments/dscp-registry.
• Additional Local Networks
Use this section to list any additional networks that are not separated from
Switchvox by a NATing router. This option is only needed if:
-Switchvox is behind a router that performs NAT.
-You wish to use soft phones from outside your network.
-There is more than one network (such as a remote office VPN) connected to the
local net.
Networks can be listed in either CIDR (e.g., 192.168.0.0/24) or netmask (e.g.,
192.168.0.0/255.255.255) notation.
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Networking
Network Proxy Settings
These settings direct all outgoing HTTP and HTTPS traffic to a web proxy.
•
•
•
•
•
Proxy Hostname or IP Address
Proxy Port
Proxy Username
Proxy Password
Custom SSL Certificates
Use these fields to add a Custom SSL Certificate. This option can be used to prevent
your browser from popping up a security warning when you access the Admin Suite.
After you have purchased an HTTPS certificate from an online certificate authority, it
can be uploaded into these sections to allow Switchvox to use it for encrypting web
traffic.
Another option for this section is if your organization creates its own web certificates
for encrypting various intranet traffic.
If you are unsure about HTTPS certificates, it is safe to leave this section blank.
Access Control
The Access Control tools lets you manage which networks have access to Switchvox
It has two tabs:
• Access Control Rules allow network access to Switchvox services based on IP
address and netmask. The default action is to deny access, so if you don’t allow a
service for a network, then the network is denied access to that service.
• Blocked IP Addresses are automatically blocked by Switchvox. This tool lets you
see what has been blocked and lets you unblock it. It also lets you change the
blocking options and search for blocked IPs.
Access Control Rules
tab
The Local Network rule by default allows all traffic for all services. The All Networks rule allows traffic for various services depending on your Switchvox configuration. You can modify the services allowed for the Local Network and All Networks
rules, but you cannot change the name or the network.
Create an Access Control Rule. Click Create Access Control Rule and enter or set
the following information for the rule:
• Rule Name. Enter a unique Rule Name.
• Network. Enter the network IP address and netmask that the rule applies to.
• Never Block IPs. Switchvox will not automatically block a network. For example,
you can select YES for Local Network to Never block IPs. For details about how
Switchvox automatically blocks IP addresses and how to change it, see “Blocked
IP Addresses Tab” page 129.
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Networking
Allowed Services. Set access to YES for each service that you want to allow the network to access.
-Web Admin Portal. The Switchvox Admin Tool Suite.
-Web User Portal. The Switchvox User Tool Suite (for extension owners).
-Admin API. API calls to Admin tools.
-User API. API calls to User tools.
-Printing. Printing to Switchvox, including printing files to be faxed.
-IMAP. The Switchvox IMAP server for voicemail and faxes.
-XMPP Server (Jabber). XMPP chat server traffic.
-XMPP Client (Jabber). XMPP chat client traffic.
-SIP. SIP traffic.
-IAX. IAX traffic.
-NTP. NTP traffic (Network Time Protocol).
-SNMP. SNMP traffic. If you want to monitor Switchvox using SNMP, make
sure this service is checked. For information about the OIDs that Switchvox
publishes, see “SNMP” page 117.
-Digium Config Server. The phone configuration server for Digium Phones.
This must be On to allow access from a remote Digium Phone (meaning, Digium Phones on an external network).
Blocked IP Addresses
Tab
Switchvox automatically blocks an IP if it is attempting to log in or register a phone
with a bad username and password. This is based on the assumption that if someone is
trying to reach Switchvox without valid login information, that person probably does
not have a legitimate reason to reach Switchvox.
• To change the options for automatic blocking, click Edit Blocking Options.
• To unblock an IP address or netmask, click Unblock on that item.
• To find an IP address that has been blocked, use the Search field. You can enter an
IP address, an extension number, or an administrator's login.
NOTE: If you have physical access to the Switchvox server, you can use the Basic
Server Function Unblock Local IPs to remove the blocks for local IP addresses.
(See “Basic Server Functions” page 130)
IP Blocking Options
The IP Blocking Options determine when Switchvox blocks an IP address from logging into or connecting to Switchvox.
Web Suite and API Logins. --
• Login attempts before lock out. If an IP address attempts to log in as a Switchvox extension or administrator, but fails this number of times, then the IP address
is locked out of Switchvox. That means the IP can reach the Switchvox Web Suite
and API, but no login information can be entered.
• Login attempts before IP block. If an IP address attempts to log in as a Switchvox extension or administrator, but fails this number of times, then the IP address
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Networking
is blocked from connecting to Switchvox. That means that Switchvox does not let
that IP display any part of the Web Suite, or access the API.
• Number of blocked IPs from the same network to trigger a netmask block. If
this number of blocked IPs are from the same network, then the /24 network is
blocked. In other words, if the first 3 sets of digits match on more than 25 blocked
IPs, then the netmask block (e.g., 216.239.51.0/24) replaces all of the original IP
blocks. That means that all of the 256 possible IP addresses in that netmask are
denied access to Switchvox.
SIP Phone Configuration and Registration
• Registration attempts before IP block. If a phone attempts to register, but fails
this number of times, then its IP address is blocked from connecting to Switchvox.
• Number of blocked IPs from the same network to trigger a netmask block. If
this number of blocked IPs are from the same network, then the /24 network is
blocked. In other words, if the first 3 sets of digits match on more than 25 blocked
IPs, then the netmask block (e.g., 216.239.51.0/24) replaces all of the original IP
blocks. That means that all of the 256 possible IP addresses in that netmask are
denied access to Switchvox.
General
• Share blocked IPs with Digium. If this is checked, it allows Switchvox to send
the blocked IP addresses to Digium. This may help us identify common IP
addresses that are being blocked.
Manage Physical
Access to the Server
If you have an appliance with an LCD Panel, you can disable other access to the appliance. You can also clear the password on the LCD Panel, in case you have forgotten it.
Disable server configuration from a local keyboard. Check this box if you do not
want anyone to be able to connect a keyboard and monitor to the Switchvox server
and make changes to the server configuration. This means that the LCD Panel is the
only way to access Switchvox's Basic Server Functions. You can set a password to
protect the use of the LCD Panel.
Clear LCD Password. Click this button to clear the password for the LCD Panel.
This means that anyone with access to the Switchvox server can use the LCD Panel to
configure Switchvox’s Basic Server Functions, without entering a password.
Basic Server
Functions
There are several functions that can or must be done at the Switchvox server; those
functions are listed here. Use the LCD Panel on your Appliance if there is one, or connect a keyboard and monitor to the Appliance.
• View System Info. This displays the URL of the Switchvox Admin Tool Suite.
• Configure Network. This lets you change your network settings. These are the
same options that you used when you first set up your Switchvox server (see Configuring Your Network Installing Switchvox). You can only do this from here at
the server.
• Reboot PBX. This reboots Switchvox. You can also do this from Server > Maintenance: System Reload (see System Reload).
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Networking
• Shutdown PBX. This shuts down Switchvox and turns off the power on the appliance. You can also do this from Server > Maintenance: System Reload (see System Reload).
• Reset HTTPS certificate. This resets your SSL certificate. You can only do this
from here at the server.
• Tech Support Access. This lets Switchvox technical support representatives log
into your Switchvox Appliance. You can also do this from Reporting > Diagnostics: Tech Support (see Technical Support).
• Restore Web Access. This restores local network access to the Web-based Switchvox Admin Tool Suite. If you turned off this access in the Local Network rule in
Access Control, you can turn it back on using this option. You can only do this
from here at the server.
• Reset Admin Passwd. This resets the main Admin password (user name ‘admin’)
to admin, which is the default password. You can only do this from here at the
server.
• Unblock Local IPs. This removes the block(s) for the local network. That means
any Blocked IPs where the first 3 sets of digits match the first 3 sets in Switchvox's
IP (e.g., a /24 CIDR). You can only do this from here at the server. (See also:
Blocked IPs.).
• Change LCD Passwd. This lets you set or change the password for the LCD
Panel. If you cannot remember your LCD password, you can clear it using the
Admin Suite (see Managing Physical Access to the Server). You can only do this
from here at the server, and it is only available if you have an Appliance with an
LCD Panel.
• Lock LCD. This immediately puts the LCD Panel in password-request mode. You
can only do this from here at the server, and it is only available if you have an
Appliance with an LCD Panel.
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Networking
Phone Networks
Phone Networks determine how a Digium Phone should behave, depending on the IP
address and netmask of the Phone as it sits on the network. If the Phone cannot get on
the network because it is trying in vain to use the wrong Phone Networks information,
use the Phone's Select Network option to select the Phone Network that matches the
network you want the phone to use.
IMPORTANT: for a remote phone (a phone that is not on the same network
as Switchvox), make sure that you have an Access Control Rules in place
(see“Create an Access Control Rule” on page 128), and that your router is
forwarding the appropriate ports to Switchvox.
The Phone Network screen lists all phone networks with their respective information.
Three actions are available for each entry: display Details, Modify, and Delete.
When creating a Phone Network, enter the following information, and click the Save
Phone Network button when finished. This lists the network on the Phone Network
screen. Modify options are the same as the Create options.
Create Phone
Network
General Settings
Name. A unique name for this network.
Network. The IP address and Netmask of the network where Digium Phones could be
located. This should be the IP address that the phone obtains and uses to identify itself
on this network. For example, this might be a subnet in your office, or the IP address
of a phone at an employee's home. (This is not a remote employee's external IP
address, which you would use in an Access Control Rule.)
Direct Port Access. Set to YES if the phone, while on this network, has access to
these ports on Switchvox:
Port 80, for getting phone firmware. (If a remote phone cannot reach your
Switchvox to get phone firmware, the phone will get the firmware from Digium's
servers).
Port 443, for getting additional files and accessing Switchvox's API.
Port 5062, for getting configuration information.
Normally a phone just uses port 5060 for all of these activities and for handling calls.
However, if set to YES, the phone uses those ports as indicated. This is useful because
those protocols are better suited for those activities. That's why the default phone network Internal is set to YES.
NTP Server. By default, Digium Phones use NTP to retrieve time settings. Unless
you have a particular reason, do not change these settings.
Primary Host (PBX)
Host Address. Switchvox’s IP address to be used by a Digium Phone located on this
network.
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Networking
The default Phone Networks entries (Internal and All Networks) are automatically
updated so that Host Address is the same as Switchvox's internal and external IP
addresses in IP Configuration. However, if you make a change to Host Address in
either one of the default entries, then changes you make in IP Configuration will not
ever affect that entry.
Alternate Host
Host Address. In the event that the Phone can no longer reach Switchvox at the PBX
Address above, the Phone will register to this IP address.
If this happens, the Phone will be in Failover Mode, and many of its Phone Applications will not be available. However, the phone should be able to make and receive
calls over this alternate host.
Port. Enter the port number. 5060 is the default port number.
Transport. Select the transport protocol, UDP or TCP. UDP is the default.
Advanced Settings
To enter advanced settings, click the Advanced Settings button. This displays the following options:
Primary Host (PBX)
Seconds until Re-registration. Enter the number of seconds until the phone re-registers to the host.
Seconds until Retry on Failure. If the phone fails to register to the host, enter the
number of seconds it should wait before it tries again.
Alternate Host
Seconds until Re-registration. Enter the number of seconds until the phone re-registers to the alternate host.
Seconds until Retry on Failure. If the phone fails to register to the alternate host,
enter the number of seconds it should wait before it tries again.
Virtual LAN Settings
Discovery Mode. Select one of these modes:
• LLDP is the default. In this case, a Digium Phone uses LLDP to discover Virtual
Local Area Network information. If the phone does not get a response from LLDP,
it will assume that there is no VLAN. (VLAN information cannot be obtained via
DHCP.)
• Manual. Use to set a specific VLAN ID:
lldp-med.policy.vlan.vid
• None. Use if you do not want the phone to attempt to use LLDP, and you do not
want the phone to use a specific VLAN ID.
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Networking
ID. Four-digit identifier for the VLAN if the VLAN discovery mode is Manual.
Voice and SIP 802.1p Priority. You can set priority values here, but if LLDP returns
specific values to the phone, the values entered here are ignored.
Quality of Service Settings
Voice DSCP and SIP DSCP. You can set priority values here, but if LLDP returns
specific values to the phone, the values entered here are ignored.
Miscellaneous Settings
Enable UDP Persist-Connection. Select NO or YES.
UDP Persist Connection Interval. Set the interval using the sliding scale. This will
attempt to keep NAT tunnel alive by pinging server on a regular interval.
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Index
Index
A
Abandoned Calls 108
Access Control 128
Access Control Rules 128
Actions Types 68
Adding a Sound 98
Adding an Action to an IVR Menu 68
Administration Account 64
Admins tools 63
Advanced Debugging 120
Aggressive Echo Cancellation 41
All Extensions 91
All Networks rule 128
Alter Caller ID 72
Alternate Host Default Option 37
Analog Phone 8
Apply Noise Reduction Filter 41
Assigned Phones and Extensions 33
Audio Codecs 49, 55
Auto-answer Switchboard Initiated Calls 15
Auto-Attendant 16
Automatic Gain Correction 42
Automatically Determine Call Progress 42
Average Entry Position 108
Average Talk Time 108
Average Wait Time for Abandoned Calls 108
Average Wait Time for Completed Calls 108
B
Backups 122
Basic Server Functions
Battery Debounce 42
Battery Threshold 42
130
C
Call Activity 105
Call API Settings 11
Call Logs 104
Call Queue extension 17
Call Recording 81
Call Recording Settings 83
Call Reports 105
Call Routing 39
Callback Extension 41
Caller ID Signaling 42
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Caller ID Start 42
Channel Groups 39
Chart Report 106, 108
Codecs 14
Common Settings 15
Completed Calls 108
Concurrent Calls Report 103
Configured Phones 36
Connection Status 113
Contacts 91
Create Block Number Rule 62
Create Transfer Call Rule 62
Creating a New Time Frame 80
Creating Notification Templates 90
Current Calls 115
Current Calls Waiting 108, 109
Custom Digit Map Timeout 38
Custom Switchboard Panels 94
Custom Voicemail Message Length 88
D
Default Fax Extension 41
Deleting a Sound 100
Deleting a Time Frame 81
Deleting Notification Templates 91
Detect Busy Signal 42
Device Types 40
BRI 41
BRI B (Bearer) Channel 41
BRI D (Data) Channel 41
E&M Signaling 41
FXO (Foreign Exchange Office) with Groundstart
Signaling 40
FXO (Foreign Exchange Office) with Loopstart signaling 40
FXO and FXS 40
FXS (Foreign Exchange Station) with Groundstart
Signaling 40
FXS (Foreign Exchange Station) with Kewlstart
Signaling 40
FXS (Foreign Exchange Station) with Loopstart
Signaling 40
PRI 40
PRI T1/E1 B (Bearer) Channel 41
PRI T1/E1 D (Data) Channel 41
DHCP Sniping 38
Diagnostics 113
135
Index
H
DID Route 60
Digit Map Options 38
Digium Addon Products 84
Digium Phone Assignment 33
Disk Usage 116
Distinctive Ringtones 92
Distinctive Ringtones List 92
DTMF Mode 14
Duplicate Extension 29
Hang up on Polarity Switch 43
Hardware Devices
List of Channels 115
Hardware Echo Cancellation 43,
I
E
Echo Cancel 42
Echo Cancel when bridging TDM channels 43
Echo Canceller Tail Length 43
Echo Canceller Type 43
Echo Reduction 121
Enable Audio Failover 52
Error Correction Mode for T.38 Faxes 51
Error Logs 120
Event Triggers 84
Extension Features 91
Extension File 28
Extension Groups 16
Extension length 38
Extension Template 28
Extension Type 28
Extensions 6
Creating 8
extensions
deleting and recreating 29
Evaluate Import 29
updating existing 29
External Contacts for Digium Phones 91
External Phones 114
External Voicemail 87
F
Fax Buffer Policy 43
fax buffers 44
Fax Detection 51
Fax Detection for this Channel Group 43
Fax Error Correction Mode 43
Fax Print Jobs 115
Fax Settings 51, 55
File Examples 28
File Information 28
Finding Product Addons 84
Finding the IP Address of your SIP Phone
FXS Channel 9
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115
IAX Providers 52
Ignore Weekends 105, 107
Ignored Phones and Extensions 34
Import Extensions 27
In Queue Call Routing 20
Incoming Caller ID Rule 62
Incoming Calls 60
Initiate T38 Reinvite After Timeout 52
Internal Directory 91
Interruptible Actions 68
Interruptible Sound 69
IP Configuration 126
IVR
Change Language 72
Check User Password 78
Concatenate Variables 76
Conditional Clause 72
Gate Keeper 71
Get Extension Status 77
Get Extension Type 77
Get Global Variable 75
Perform Math 75
Play DTMF Tones 74
Record Digits 74
Remove Distinctive Ring Hint 78
Say A Number 74
Say Date/Time 74
Say Digits/Letters 74
Send an Email 70
Send Call Values to URL 76
Set Distinctive Ring Hint 78
Set Global Variable 75
Set Variable 75
Time Based Clause 72
XML and Returned Values 76
IVR Extension 17
IVR Extensions 16
IVR Menu Entry Point 17
IVR Options 79
L
Language and Locale
9
136
Index
Language for Sound 69
Log of Call Activity 104
Login Actions 10
Longest Queue Length 108
Longest Wait Time for Abandoned Calls 108
Longest Wait Time for Completed Calls 108
M
Mailbox Quota
FAX 10
Voicemail 10
Maintenance Tools 121
Making Phones Visible to Switchvox 37
Manage Announcements 20
Managing Languages 101
Media 96
Member Status 109
Merge Sounds 71
Modifying a Time Frame 81
Modifying an Existing Action 68
Modifying Notification Templates 90
Modifying/Playing a Sound 99
Moving a Sound into a Folder 100
Music On Hold 96
N
Network Proxy Settings 128
Network Specific Facility Code
Networking 126
Notification 89
Notification Templates 89
Numeric Password 9
Change Custom Settings 16
Change Profile Information 15
Create Call Rules 15
Use Digium Phone Apps 16
permissions
Change Digium Phone Settings 16
Use Calling Features 16
Use Distinctive Ringtones 16
Use PBX Features 15
Use Phone Feature Pack Phone Features 16
Use Voicemail Features 15
Phone Assignment Options 34
Phone Discovery Options 38
Phone Feature Packs 34, 124
Phone Model Not Reported 36
Phone NAT Traversal 15
Phone Networks 132
Phone Password 15
Phone Settings
Common 13
Digium 11
Polycom 14
Phone Setup 36
Phone Setup Options 37
Phone States 114
Play Recorded Sound 69
Play Sound 69
Play Sound From URL 71
Profile Information 9
Profile Picture 10
44
Q
O
Only Accept Calls from the PBX 37
Outbound SMTP Server 89
Outbound SMTP Username and Password
Outgoing Call Diagnostics 60
Outgoing Call Rule 58
Outgoing Call Rules 16
Outgoing Calls 58
Output to .xls File 104, 106
Output to xml 106, 108
P
Permissions
Switchvox Admin Reference Manual
89
Queue Extension 17
Queue Logs 107
Queue Member Settings 18
Queue Members 17
Queue Name 17
Queue Reports 107
Queue Setup 17
Queue Statistics 108
Queue Status 108
Quick-Create Assignments 34
R
Real-time Information for Each Call Queue 108
Record Sound 69
Redirected Calls 108
Report Parameters 110
Ringing Strategy 17
Route to IVR Menu 17
137
Index
RSA Key
56
S
Sample DTD 76
Schedule Information 110
Scheduled Reports 110
Scheduling a Call Recording 81
Server Information 116
Advanced 116
Single DID Route Settings 61
SIP Phone or SIP Adapter for Analog Phone 8
SIP Phones 114
SIP Providers 46
SMTP Settings 89
SNMP 117
Sound Manager 98
Sound Manager folder 69
Sound packs 100
Sound Prompt Language 11
Sound to Play 69
Store Recorded Sound in Sound Manager 70
SugarCRM® 93
Switchboard Panels 92
Switchvox OIDs 117
System Clock 125
System Reload 125
T
Technical Support 120
Telephony Cards 121
Time Frames 80
U
Unassigned Phones 33
Unconfigured phones 36
Unknown phones 35
Upload Recorded Sound 70
V
Video Codecs 49
Viewing & Playing Recorded Calls 82
Virtual Extension 9
Voicemail 87
Voicemail Notification Email “From” Address 89
Voicemail Routing 87
VOIP Providers 113
VoIP Providers 46
X
XML Response
Switchvox Admin Reference Manual
86
138
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