Application Notes for Configuring SIP Trunking Using Verizon

Application Notes for Configuring SIP Trunking Using Verizon
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring SIP Trunking Using
Verizon Business IP Trunk SIP Trunk Service Offer and
Avaya IP Office Release 6 – Issue 1.0
Abstract
These Application Notes describe a sample configuration using Session Initiation Protocol
(SIP) trunking between the Verizon Business IP Trunk SIP Trunk Service Offer and an Avaya
IP Office solution. In the sample configuration, the Avaya IP Office solution consists of an
Avaya IP Office 500 Release 6 Preferred Edition, Avaya Voicemail Pro, Avaya IP Office
Softphone, and Avaya H.323, digital, and analog endpoints.
The Verizon Business IP Trunk service offer referenced within these Application Notes is
designed for business customers. The service enables local and long distance PSTN calling
via standards-based SIP trunks directly, without the need for additional TDM enterprise
gateways or TDM cards and the associated maintenance costs.
Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions.
Testing was conducted in the Avaya
Interoperability Test Lab, utilizing a Verizon Business Private IP (PIP) circuit connection to
the production Verizon Business IP Trunking service.
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Table of Contents
Table of Contents ............................................................................................................................ 2
1. Introduction ............................................................................................................................. 3
1.1. Known Limitations ......................................................................................................... 5
2. Equipment and Software Validated ........................................................................................ 6
3. Avaya IP Office Configuration ............................................................................................... 6
3.1. Physical, Network, and Security Configuration ............................................................. 7
3.2. Licensing ......................................................................................................................... 9
3.3. System Settings ............................................................................................................. 11
3.3.1.
System Tab............................................................................................................ 11
3.3.2.
LAN Settings ........................................................................................................ 12
3.3.3.
Voicemail .............................................................................................................. 14
3.3.4.
System Telephony Configuration ......................................................................... 14
3.3.5.
System Twinning Configuration ........................................................................... 15
3.4. SIP Line ........................................................................................................................ 15
3.4.1.
SIP Line - SIP Line Tab ........................................................................................ 16
3.4.2.
SIP Line - SIP URI Tab ........................................................................................ 17
3.4.3.
SIP Line - VoIP Tab ............................................................................................. 18
3.5. Users, Extensions, and Hunt Groups ............................................................................ 19
3.5.1.
User 201 ................................................................................................................ 19
3.5.2.
User 30026 ............................................................................................................ 22
3.5.3.
Hunt Groups .......................................................................................................... 25
3.6. Short Codes ................................................................................................................... 28
3.7. Incoming Call Routes ................................................................................................... 30
3.8. ARS and Alternate Routing .......................................................................................... 33
3.9. Privacy / Anonymous Calls .......................................................................................... 35
3.10.
SIP Options Frequency ............................................................................................. 36
3.11.
Saving Configuration Changes to IP Office ............................................................. 37
4. Verizon Business Configuration ........................................................................................... 38
5. Verifications .......................................................................................................................... 39
5.1. Verification Summary................................................................................................... 39
5.2. System Status Application ............................................................................................ 40
5.3. System Monitor Application ......................................................................................... 43
5.4. IP Office Softphone Application .................................................................................. 47
5.5. Mobile Twinning Application....................................................................................... 53
5.6. Outbound Anonymous / Private Calls .......................................................................... 55
5.7. SIP OPTIONS ............................................................................................................... 55
5.8. DiffServ Markings by IP Office ................................................................................... 56
6. Support .................................................................................................................................. 57
6.1. Avaya ............................................................................................................................ 57
6.2. Verizon .......................................................................................................................... 57
7. Conclusion ............................................................................................................................ 57
8. References ............................................................................................................................. 57
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1. Introduction
These Application Notes describe a sample configuration using Session Initiation Protocol (SIP)
trunking between the Verizon Business IP Trunk SIP Trunk Service Offer and an Avaya IP
Office solution. In the sample configuration, the Avaya IP Office solution consists of an Avaya
IP Office 500 Release 6 Preferred Edition, Avaya Voicemail Pro, Avaya IP Office Softphone,
and Avaya H.323, digital, and analog endpoints.
Customers using Avaya IP Office with the Verizon Business IP Trunk SIP Trunk service are able
to place and receive PSTN calls via the SIP protocol. The converged network solution is an
alternative to traditional PSTN trunks such as ISDN-PRI.
Verizon Business IP Trunk service offer can be delivered to the customer premise via either a
Private IP (PIP) or Internet Dedicated Access (IDA) IP network terminations. Although the
configuration documented in these Application Notes used Verizon’s IP Trunk service
terminated via a PIP network connection, the solution validated in this document applies equally
to IP Trunk services delivered via IDA service terminations.
For more information on the Verizon Business IP Trunking service, including access
alternatives, visit http://www.verizonbusiness.com/us/products/voip/trunking/
Table 1 shows the mapping of Verizon-provided DID numbers to IP Office users, groups, or
functions. The associated IP Office configuration is shown in Section 3. Selected verifications
are illustrated in Section 5.
Verizon Provided DID
732-945-0228
Avaya IP Office Destination
Auto-Attendant on
Voicemail Pro
732-945-0229
x30026
732-945-0231
x201
732-945-0232
732-945-0233
x203
x209
732-945-0234
Voicemail Collect on
Voicemail Pro
x30025
“200 Main” Hunt Group
30200 Hunt Group
732-945-0235
732-945-0236
732-945-0237
Notes
See Section 3.6
Avaya 1616 Telephone, or
Avaya IP Office Softphone
logged in as x30026
Digital Telephone with
Mobile Twinning Active
Digital Telephone
Analog telephone or Fax
machine, see Section 1.1
See Section 3.6, also used for
Voicemail Pro Callback
Avaya 1616 IP Telephone
Collective Ring Hunt Group
Sequential Ring Hunt Group
Table 1: Verizon DID to IP Office Mappings
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Figure 1 illustrates an example Avaya IP Office solution connected to the Verizon Business IP
Trunk SIP Trunk service. The Avaya equipment is located on a private IP subnet. An enterprise
edge router provides access to the Verizon Business IP Trunk service network via a Verizon
Business T1 circuit. This circuit is provisioned for the Verizon Business Private IP (PIP) service.
The Verizon network configuration is identical to the configuration described in reference [CMVZIPT].
In the sample configuration, IP Office receives traffic from the Verizon Business IP Trunk
service on port 5060 and sends traffic to port 5071, using UDP for network transport, as required
by the Verizon Business IP Trunk service. As shown in Table 1, the Verizon Business IP Trunk
service provided Direct Inward Dial (DID) numbers. These DID numbers were mapped to IP
Office destinations via Incoming Call Routes.
Verizon Business used FQDN pcelban0001.avayalincroft.globalipcom.com. The Avaya IP
Office environment was assigned FQDN adevc.avaya.globalipcom.com by Verizon Business.
Figure 1: Avaya IP Office with Verizon IP Trunk SIP Trunk Service
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1.1. Known Limitations
Interoperability testing of the sample configuration was completed with successful results, with
the successful verifications summarized in Section 5. The following observations were noted:
1. For Compression Mode, either G.729a 8K CS-ACELP or G.711 ULAW 64K can be
selected for voice calls. Note that T.38 fax is not supported by the Verizon IP Trunk SIP
Trunk Service, and T.38 fax is the only fax method supported using SIP Lines on IP
Office. Although the Verizon Business IP Trunk Service does not support T.38 fax, and
T.38 fax is the only fax method supported by IP Office for SIP Lines, calls were
nevertheless made to and from an analog fax machine connected to IP Office. Fax calls
may succeed using G.711 but cannot be guaranteed. Although not supported, if fax is to
be attempted, G.711 ULAW 64K can be selected as the only allowed codec on the SIP
Line. Alternatively, if G.729a is desired for voice calls and G.711 for fax calls, the ReInvite Supported option for the SIP Line can be checked to allow re-negotiation to
G.711 for a call involving a fax machine that begins at G.729a.
2. When a call is put on hold by an IP Office user, there is no indication sent via SIP
messaging to Verizon. This is transparent to the users that are party to the call.
3. With the configuration described in these Application Notes, when a call uses the IP
Office Mobile Twinning feature, and the call is delivered via the Verizon Business IP
Trunk service to a PSTN telephone (e.g., a mobile telephone), the calling number
displayed on the PSTN telephone (e.g., mobile phone) is the actual caller’s calling party
number. However, for other types of call diversion (e.g., call forwarding), the calling
number displayed on the diverted-to PSTN telephone is the DID associated with the
Avaya IP Office telephone (e.g., the forwarding user), and not the actual caller’s calling
party number. IP Office will populate the Diversion header for Mobile Twinning, but not
for other forms of diversion such as call forwarding.
4. When using the IP Office Softphone, inbound PSTN calls from the Verizon Business IP
Trunk service to the IP Office Softphone may negotiate to the G.711MU codec, even if
the SIP Line configuration lists G.729a first on the VoIP tab (as shown in Section 3.4.3).
Specifically, if the IP Office Softphone user has logged in with the “IP Office: Default”
profile, and the Automatic Codec Preference parameter on the System  Telephony
tab is set to “G.711 ULAW 64K”, an inbound call from the Verizon Business IP Trunk
service to the IP Office Softphone will use G.711MU. The IP Office Softphone user can
log in with the “IP Office: Low Bandwidth” profile to ensure use of G.729a for both
inbound and outbound calls via the SIP Line to Verizon Business.
5. Although ARS alternate routing was tested successfully, there is no capability to control
the timing of the alternate routing. IP Office will wait a short fixed amount of time (e.g.,
roughly 5 seconds) for a response after sending INVITE to a primary SIP Line. If no
response is received, route advance to a configured alternate route occurs after
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approximately 5 seconds. The IP Office product team expects to provide configurable
control of this timing in a future release.
6. For inbound PSTN calls to IP Office, if Display information is provided by Verizon in
the To Header, this display information is sent by IP Office in the 200 OK as the User
portion of the Contact in the Contact header, rather than the DID number configured for
the answering user. In the sample configuration, this does not cause a user-perceivable
problem, but this is an anomaly that the IP Office product team will correct in a future
release.
7. IP Office Release 6 does not support DNS SRV records. Static provisioning of the
Verizon Business SIP IP Address and port are shown in these Application Notes.
2. Equipment and Software Validated
Table 2 shows the equipment and software used in the sample configuration.
Equipment
Software
Avaya IP Office 500
Release 6.0 (6.0.8) (Preferred Edition)
Avaya IP Office Manager
Release 8.0 (8.0.8) (Preferred Edition)
Avaya IP Office Voicemail Pro
Release 6.0.16
Avaya IP Office Voicemail Pro Client
Version 6.0 (16)
Avaya 1600-Series Telephones (H.323)
Avaya 2400-Series and 5400-Series Digital
Telephones
Avaya IP Office Softphone
Brother Intellifax 1360 (analog fax)
Release 1.2.2
REL: 6.00 (downloaded from IP Office)
Release 3.0 (56516)
N/A
Table 2: Equipment and Software Tested
3. Avaya IP Office Configuration
IP Office is configured via the IP Office Manager program. For more information on IP Office
Manager, consult reference [IPO-MGR]. From the IP Office Manager PC, select Start →
Programs → IP Office → Manager to launch the Manager application. A screen that includes
the following in the center will be displayed:
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Open the IP Office configuration, either by reading the configuration from the IP Office server,
or from file. The appearance of the IP Office Manager can be customized using the View menu.
In the screens presented in this section, the View menu was configured to show the Navigation
pane on the left side, the Group pane in the center, and the Details pane on the right side.
3.1. Physical, Network, and Security Configuration
This section describes attributes of the sample configuration, but is not meant to be prescriptive.
Consult reference [IPO-INSTALL] for more information on the topics in this section.
In the sample configuration, looking at the IP Office 500 from left to right, the first module is
blank (i.e., no module is inserted). The next slots from left to right contain a VCM64, a Digital
station module, and a “Phone8” analog module. The VCM64 is a Voice Compression Module
supporting VoIP codecs. The Digital module allows connection of Avaya 5400-Series and
Avaya 2400-Series Digital telephones. Referring to Figure 1, the Avaya 5410 telephone with
extension 201 is connected to port 1 of the Digital module, and the Avaya 2410 telephone with
extension 203 is connected to port 3 of the Digital module. The “Phone8” module allows
connection of analog devices such as simple analog telephones or fax machines. In the testing of
the sample configuration, an analog telephone or a fax machine is connected to port 1 of the
“Phone8” module.
The following screen shows the modules in the IP Office used in the sample configuration. To
access such a screen, select Control Unit in the Navigation pane. The modules appear in the
Group pane. In the screen below, IP 500 is selected in the Group pane, revealing additional
information about the IP 500 in the Details pane.
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In the sample configuration, the IP Office LAN1 port is physically connected to the local area
network switch at the IP Office customer site. The default gateway for this network is 1.1.1.1.
To add an IP Route in IP Office, right-click IP Route from the Navigation pane, and select New.
To view or edit an existing route, select IP Route from the Navigation pane, and select the
appropriate route from the Group pane. The following screen shows the Details pane with the
relevant default route using Destination LAN1.
To facilitate use of Avaya IP Office Softphone, https was enabled in the sample configuration.
To check whether https is enabled, navigate to File  Advanced  Security Settings. A
screen such as the following is presented. Log in with the appropriate security credentials.
After logging in, select System from the Navigation pane and the appropriate IP Office system
from the Group pane. In the Details pane, select the System Details tab. Verify that Allow
HTTPS is checked. If not, check the box, click OK, and heed the on-screen prompts and
warnings. Note that this action may be service disrupting.
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When complete, select File  Configuration to return to configuration activities.
3.2. Licensing
The configuration and features described in these Application Notes require the IP Office system
to be licensed appropriately. If a desired feature is not enabled or there is insufficient capacity,
contact an authorized Avaya sales representative.
To verify that there is a SIP Trunk Channels License with sufficient capacity, click License in
the Navigation pane and SIP Trunk Channels in the Group pane. Confirm a valid license with
sufficient “Instances” (trunk channels) in the Details pane.
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If Avaya IP Telephones will be used, verify the Avaya IP endpoints license. Click License in
the Navigation pane and Avaya IP endpoints in the Group pane. Confirm a valid license with
sufficient “Instances” in the Details pane.
A similar process can be used to check the license status for other desired features. For example,
the following screen shows the availability of a valid license for Mobility features. In the sample
configuration, various mobility features including Mobile Twinning are used.
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The following screen shows the availability of a valid license for Power User features. In the
sample configuration, the user with extension 30026 will be configured as a “Power User” and
will be capable of using the Avaya IP Office Softphone.
3.3. System Settings
This section illustrates the configuration of system settings. Select System in the Navigation
pane to configure these settings. The subsection order corresponds to a left to right navigation of
the tabs in the Details pane for System settings.
3.3.1. System Tab
With the proper system name selected in the Group pane, select the System tab in the Details
pane. The following screen shows a portion of the System tab. The Name field can be used for
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a descriptive name of the system. In this case, the MAC address is used as the name. The
Avaya HTTP Clients Only and Enable SoftPhone HTTP Provisioning boxes are checked to
facilitate Avaya IP Office Softphone usage.
3.3.2. LAN Settings
In the sample configuration, LAN1 was used to connect the IP Office to the enterprise network.
Other LAN choices (e.g., LAN2) may also be used. To view or configure the IP Address of
LAN1, select the LAN1 tab followed by the LAN Settings tab. As shown in Figure 1, the IP
Address of the IP Office, known to Verizon Business, is 1.1.1.2. Other parameters on this screen
may be set according to customer requirements.
Select the VoIP tab as shown in the following screen. The SIP Trunks Enable box must be
checked to enable the configuration of SIP trunks to Verizon Business. The H323 Gatekeeper
Enable box is checked to allow the use of Avaya IP Telephones using the H.323 protocol, such
as the Avaya 1600-Series Telephones used in the sample configuration. The SIP Registrar
Enable box is checked to allow Avaya IP Office Softphone usage.
If desired, the RTP Port Number Range can be customized to a specific range of receive ports
for the RTP media paths from Verizon Business to IP Office. That is, for SIP Trunk calls to and
from Verizon Business, the SIP protocol exchanges will result in Verizon Business sending RTP
media to IP Office using a UDP port in the configurable range shown below.
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If desired, IP Office can be configured to mark the Differentiated Services Code Point (DSCP) in
the IP Header with specific values to support Quality of Service policies. In the sample
configuration shown below, IP Office will mark SIP signaling with a value associated with
“Assured Forwarding” using DSCP decimal 28 (SIG DSCP parameter). IP Office will mark the
RTP media with a value associated with “Expedited Forwarding” using DSCP decimal 46
(DSCP parameter). This screen enables flexibility in IP Office DiffServ markings (RFC 2474)
to allow alignment with network routing policies, which are outside the scope of these
Application Notes. Other parameters on this screen may be set according to customer
requirements.
Select the Network Topology tab as shown in the following screen. For Public IP Address,
enter the Avaya IP Office LAN1 IP address. Set the Public Port to 5060. In the sample
configuration, the Firewall/NAT Type is set to “Open Internet”, and the Binding Refresh Time
is set to 60 seconds. Later, the SIP Line will be configured to use the network topology
information for LAN1. Click the OK button.
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Note: The Firewall/NAT Type parameter may need to be different, depending on the type of
firewall or Network Address Translation device used at the customer premise.
3.3.3. Voicemail
To view or change voicemail settings, select the Voicemail tab as shown in the following screen.
The settings presented here simply illustrate the sample configuration and are not intended to be
prescriptive. The Voicemail Type in the sample configuration is “Voicemail Lite/Pro”. Other
Voicemail types may be used. The Voicemail IP Address in the sample configuration is
1.1.1.63, the IP Address of the PC running the Voicemail Pro software, as shown in Figure 1. In
the sample configuration, the “Callback” application of Avaya Voicemail Pro was used to allow
Voicemail Pro to call out via the SIP Line to Verizon Business when a message is left in a voice
mailbox. The SIP Settings shown in the screen below enable IP Office to populate the SIP
headers for an outbound “callback” call from Voicemail Pro, similar to the way the fields with
these same names apply to calls made from telephone users (e.g., see Section 3.5). Other
parameters on this screen may be set according to customer requirements.
3.3.4. System Telephony Configuration
To view or change telephony settings, select the Telephony tab and Telephony sub-tab as
shown in the following screen. The settings presented here simply illustrate the sample
configuration and are not intended to be prescriptive. In the sample configuration, the Inhibit
Off-Switch Forward/Transfer box is unchecked so that call forwarding and call transfer to
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PSTN destinations via the Verizon Business IP Trunk service can be tested. The Companding
Law parameters are set to “ULAW” as is typical in North American locales. Other parameters
on this screen may be set according to customer requirements.
3.3.5. System Twinning Configuration
To view or change Twinning settings, select the Twinning tab as shown in the following screen.
The Send original calling party information for Mobile Twinning box is not checked in the
sample configuration, and the Calling party information for Mobile Twinning is left blank.
With this configuration, and related configuration of “Diversion header” on the SIP Line
(Section 3.4), the true identity of the caller can be presented to the twinning destination (e.g., a
user’s mobile phone) when a call is twinned out via the Verizon Business IP Trunk service. An
example Wireshark trace for a call using Diversion Header with Mobile Twinning is shown in
Section 5.
3.4. SIP Line
To add a new SIP Line, right click on Line in the Navigation pane, and select New  SIP Line.
A new Line Number will be assigned automatically. To edit an existing SIP Line, click Line in
the Navigation pane, and the SIP Line to be configured in the Group pane.
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3.4.1. SIP Line - SIP Line Tab
The SIP Line tab in the Details pane is shown below for Line Number 7, used for the Verizon
Business IP Trunk service. The ITSP IP Address is set to the IP Address provided by Verizon
Business. As shown in Figure 1, this IP Address is 172.30.209.21. In the Network
Configuration area, UDP is selected as the Layer 4 Protocol, and the Send Port is set to the
port number provided by Verizon Business. As shown in Figure 1, this port is 5071 in the
sample configuration. Note that IP Office does not support DNS SRV for lookup of the SIP IP
Address and port. The Use Network Topology Info parameter is set to “LAN 1”. This
associates the SIP Line with the parameters in the System  LAN1  Network Topology tab.
By default, Check OOS is checked. In the sample configuration, IP Office will use the SIP
OPTIONS method to periodically check the SIP Line. The time between SIP OPTIONS sent by
IP Office will use the Binding Refresh Time for LAN1, which has been set to 60 seconds as
shown in Section 3.3.2. See Section 3.10 for additional information related to configuring the
periodicity of SIP OPTIONS.
The Send Caller ID parameter is set to “Diversion Header”. With this setting and the related
configuration in Section 3.3.5 and Section 3.9, IP Office will include the Diversion Header for
calls that are directed via Mobile Twinning out the SIP Line to Verizon. The Diversion Header
will contain the number associated with the Twinning user, allowing Verizon to admit the call,
and the From Header will be populated with the true calling party identity, allowing the twinning
destination (e.g., mobile phone) to see the true caller id. The Call Routing Method can retain
the default “Request URI” setting, or may be changed to “To Header” as shown, to match
Incoming Call Routes based on the contents of the “To Header”. In the sample configuration,
both approaches were tested successfully. Click OK (not shown).
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3.4.2. SIP Line - SIP URI Tab
Select the SIP URI tab. To add a new SIP URI, click the Add… button. In the bottom of the
screen, a New Channel area will be opened. To edit an existing entry, click an entry in the list at
the top, and click the Edit… button. In the bottom of the screen, the Edit Channel area will be
opened. In the example screen below, a previously configured entry is edited. “Use Internal
Data” is selected for the Local URI, Contact, and Display Name. Information configured on
the SIP Tab for individual users will be used to populate the SIP headers. The Registration
parameter is set to the default “0: <None>” since Verizon Business IP Trunk service does not
require registration. The Incoming Group parameter, set here to 7, will be referenced when
configuring Incoming Call Routes to map inbound SIP trunk calls to IP Office destinations in
Section 3.7. The Outgoing Group parameter, set here to 8, will be used for routing outbound
calls to Verizon via the Short Codes (Section 3.6) or ARS configuration (Section 3.8). The Max
Calls per Channel parameter, configured here to 10, sets the maximum number of simultaneous
calls that can use the URI before IP Office returns busy to any further calls. Click OK.
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In the sample configuration, the single SIP URI shown above was sufficient to allow incoming
calls for Verizon DID numbers destined for specific IP Office users or IP Office hunt groups.
The calls are accepted by IP Office since the incoming number will match the SIP Name
configured for the user or hunt group that is the destination for the call. For service numbers,
such as a DID number routed directly to voicemail, or a DID number routed to an auto-attendant
service on Voicemail Pro, the DID numbers that IP Office should admit can be entered into the
Local URI and Contact fields instead of “Use Internal Data”.
The following shows the SIP URI tab for SIP Line 7 after the SIP URIs corresponding to Voice
mail (732-945-0234) and Auto-attendant (732-945-0228) have been added.
3.4.3. SIP Line - VoIP Tab
Select the VoIP tab. In the sample configuration, the Compression Mode was configured using
the Advanced button, allowing an explicit ordered list of codecs to be specified. Place a check
mark next to the G.729(a) 8K CS-ACELP and G.711 ULAW 64K codecs to cause IP Office to
include both G.729a and G.711MU in the Session Description Protocol (SDP), in that order. The
DTMF Support parameter can remain set to the default value “RFC2833”. The Re-invite
Supported parameter can be checked to allow for codec re-negotiation in cases where the target
of an incoming call or transfer does not support the codec originally negotiated on the trunk. The
Use Offerer’s Preferred Codec parameter can be left at the default unchecked setting. In the
sample configuration, Verizon also preferred the G.729a codec in SDP, while also allowing the
G.711MU codec. The IP Office configuration shown below matches these Verizon preferences.
In the course of testing, the IP Office configuration was varied such that G.711MU was the
preferred or only codec listed, and G.711MU calls were also successfully verified. Since the
Verizon Business IP Trunk service does not support T.38 fax, the Fax Transport Support
parameter is not checked, and the T38 Fax tab need not be visited. Since the Verizon Business
IP Trunk service does not require registration, the SIP Credentials tab need not be visited.
Click OK (not shown).
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3.5. Users, Extensions, and Hunt Groups
In this section, examples of IP Office Users, Extensions, and Hunt Groups will be illustrated. In
the interests of brevity, not all users and extensions shown in Figure 1 will be presented, since
the configuration can be easily extrapolated to other users. To add a User, right click on User in
the Navigation pane, and select New. To edit an existing User, select User in the Navigation
pane, and select the appropriate user to be configured in the Group pane.
3.5.1. User 201
The following screen shows the User tab for User 201. As shown in Figure 1, this user
corresponds to the digital telephone 5410.
The following screen shows the SIP tab for User 201. The SIP Name and Contact parameters
are configured with the DID number of the user, 7329450231. These parameters configure the
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user part of the SIP URI in the From header for outgoing SIP trunk calls, and allow matching of
the SIP URI for incoming calls, without having to enter this number as an explicit SIP URI for
the SIP Line. The SIP Display Name (Alias) parameter can optionally be configured with a
descriptive name. If all calls involving this user and a SIP Line should be considered private,
then the Anonymous box may be checked to withhold the user’s information from the network.
See Section 3.6 for a method of using a short code (rather than static user provisioning) to place
an anonymous call.
From Figure 1, note that user 201 will use the Mobile Twinning feature. The following screen
shows the Mobility tab for User 201. The Mobility Features and Mobile Twinning boxes are
checked. The Twinned Mobile Number field is configured with the number to dial to reach the
twinned mobile telephone, in this case 917326870755. Other options can be set according to
customer requirements.
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The following screen shows the Extension information for this user. To view, select Extension
from the Navigation pane, and the appropriate extension from the Group pane. As stated in
Section 3.1, the Avaya 5410 telephone user with extension 201 is connected to port 1 of the
digital module.
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3.5.2. User 30026
The following screen shows the User tab for User 30026. This user corresponds to an Avaya
1616 IP Telephone that will be granted “Power User” and Avaya IP Office Softphone features.
The Profile parameter is set to “Power User”. The Enable Softphone box is checked, along
with other advanced capabilities.
Like the user with extension 201, the SIP tab for the user with extension 30026 is configured
with a SIP Name and Contact specifying the user’s Verizon Business DID number.
The following screen shows the Voicemail tab for the user with extension 30026. The
Voicemail On box is checked, and a voicemail password can be configured using the Voicemail
Code and Confirm Voicemail Code parameters. In the verification of these Application Notes,
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incoming calls from the Verizon Business IP Trunk to this user were redirected to Voicemail Pro
after no answer. Voicemail messages were recorded and retrieved successfully. Voice mail
navigation and retrieval were performed locally and from PSTN telephones, to test DTMF using
RFC 2833, and to test assignment of a Verizon DID number to the “Voicemail Collect” feature
(e.g., via the *17 short code shown in Section 3.6).
Select the Supervisor Settings tab as shown below. To allow hot desking, enter a Login Code.
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Select the Call Settings tab as shown below. Check the Call Waiting On box to allow an IP
Office Softphone logged in as this extension to have multiple call appearances (e.g., necessary
for call transfer).
The following screen shows the Source Numbers tab for the user with extension 30026.
Although the Voicemail Pro configuration is beyond the scope of these Application Notes, the
“Callback” feature has been enabled on Voicemail Pro for this user, and the Source Number
“P917326870755” has been previously added. With this configuration, when a message is left in
this user’s Voicemail Pro mailbox, a “callback” call will be initiated to “917326870755”. The
callback call will be sent to Verizon via SIP Line 7, and the From and Contact headers in the SIP
INVITE will be populated with the information configured in the System  Voicemail tab
shown in Section 3.3.3. It is possible (and more typical) for the end user to configure callback
numbers via the Voicemail Pro Telephony User Interface, rather than the administrator
configuring a callback number via the “P” Source Number in IP Office, as shown below.
To add a new Source Number, Press the Add… button to the right of the list of any previously
configured Source Numbers. To edit an existing Source Number, select the Source Number
from the list, and click Edit… When finished, click OK.
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The following screen shows the Extension information for this user, simply to illustrate the VoIP
tab available for an IP Telephone. To view, select Extension from the Navigation pane, and the
appropriate extension from the Group pane. Select VoIP in the Details pane.
3.5.3. Hunt Groups
During the verification of these Application Notes, users could also receive incoming calls as
members of a hunt group. To configure a new hunt group, right-click HuntGroup from the
Navigation pane, and select New. To view or edit an existing hunt group, select HuntGroup
from the Navigation pane, and the appropriate hunt group from the Group pane.
The following screen shows the Hunt Group tab for hunt group 200. This hunt group was
configured to contain the two digital telephones x201 and x203 in Figure 1. These telephones
will both ring when the hunt group number is called, due to the Ring Mode setting “Collective”.
Click the Edit button to change the User List.
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The following screen shows the SIP tab for hunt group 200. The SIP Name and Contact are
configured with Verizon DID 7329450236. Later, in Section 3.7, an Incoming Call Route will
map 7329450236 to this hunt group.
The following screen shows the Hunt Group tab for another hunt group 30200. This hunt group
was configured to contain the two IP telephones x30025 and x30026 in Figure 1. These
telephones will ring sequentially when the hunt group number is called, due to the Ring Mode
setting “Sequential”. That is, extension 30025 will ring first. If unanswered, extension 30026
will ring, and so on (for larger groups).
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The following screen shows the SIP tab for hunt group 30200. The SIP Name and Contact are
configured with Verizon DID 7329450237. Later, in Section 3.7, an incoming call route will
map this same Verizon DID number to this hunt group.
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3.6. Short Codes
In this section, various examples of IP Office short codes will be illustrated. To add a short
code, right click on Short Code in the Navigation pane, and select New. To edit an existing
short code, click Short Code in the Navigation pane, and the short code to be configured in the
Group pane.
In the screen shown below, the short code “9N;” is illustrated. The Code parameter is set to
“9N;”. The Feature parameter is set to “Dial”. The Telephone Number parameter is set to
“N@Domain Name or IP Address of Verizon Business IP Trunk Service” with the text string
beginning with @ in quotes. Below, the Verizon provided domain shown in Figure 1 is
configured. The Telephone Number field is used to construct the Request URI and To Header
in the outgoing SIP INVITE message. The Line Group ID parameter is set to 8, matching the
number of the Outgoing Group configured on the SIP URI tab of SIP Line 7 to Verizon
Business (Section 3.4).
This simple short code will allow an IP Office user to dial the digit 9 followed by any telephone
number, symbolized by the letter N, to reach the SIP Line to Verizon business. N can be any
number such as a 10-digit number, a 1+10 digit number, a toll free number, directory assistance
(e.g., 411), etc. This short code approach has the virtue of simplicity, but does not provide for
alternate routing or an awareness of end of user dialing. When a users dial 9 plus the number, IP
Office must wait for an end of dialing timeout before sending the SIP INVITE to Verizon
Business. Click the OK button (not shown).
Optionally, add or edit a short code that can be used to access the SIP Line anonymously. In the
screen shown below, the short code “8N;” is illustrated. This short code is similar to the “9N;”
short code except that the Verizon IP Address rather than the domain is entered in the Telephone
Number field. This is done for variety; either method can be used. The Telephone Number
field begins with the letter W, which means “withhold the outgoing calling line identification”.
In the case of the SIP Line to Verizon documented in these Application Notes, when a user dials
8 plus the number, IP Office will include the user’s telephone number in the P-Asserted-Identity
(PAI) header along with “Privacy: Id”. Verizon will allow the call due to the presence of a valid
DID in the PAI header, but will prevent presentation of the caller id to the called PSTN
destination. See Section 5 for an example verification with Wireshark trace.
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The simple “9N;” and “8N;” short codes illustrated previously do not provide a means of
alternate routing if the primary Verizon SIP line is out of service or temporarily not responding.
When alternate routing options and/or more customized analysis of the digits following the short
code are desired, the Automatic Route Selection (ARS) feature may be used. In the following
example screen, the short code 7N is illustrated for access to ARS. When the IP Office user dials
7 plus any number N, rather than being directed to a specific Line Group Id, the call is directed
to Line Group ID “50: Main”, configurable via ARS. See Section 3.8 for example ARS route
configuration for “50: Main” as well as a backup route.
The following screen illustrates a short code that acts like a feature access code rather than a
means to access a SIP Line. In this case, the Code “*17” is defined for Feature “Voicemail
Collect”. This short code will be used as one means to allow a Verizon DID to be programmed
to route directly to voice messaging, via inclusion of this short code as the destination of an
Incoming Call Route. See Section 3.7.
The following screen illustrates another short code that acts like a feature access code rather than
a means to access a SIP Line. In this case, the Code “*99” is defined for Feature “Voicemail
Collect” and Telephone Number “Attendant”. This short code will be used as a means to allow a
Verizon DID to be programmed to route directly to a Voicemail Pro “Attendant” (auto-attendant)
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application, via inclusion of this short code as the destination of an Incoming Call Route. See
Section 3.7.
3.7. Incoming Call Routes
In this section, IP Office Incoming Call Routes are illustrated. Each Incoming Call Route will
map a specific Verizon Business DID number to a destination user, group, or function on IP
Office. To add an incoming call route, right click on Incoming Call Route in the Navigation
pane, and select New. To edit an existing incoming call route, select Incoming Call Route in
the Navigation pane, and the appropriate incoming call route to be configured in the Group pane.
In the screen shown below, the incoming call route for Incoming Number “7329450231” is
illustrated. The Line Group Id is 7, matching the Incoming Group field configured in the SIP
URI tab for the SIP Line to Verizon Business in Section 3.4.
Select the Destinations tab. From the Destination drop-down, select the extension to receive
the call when a PSTN user dials 7329450231. As shown in Table 1, 7329450231 is the DID
number associated with IP Office user extension 201.
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In the screen shown below, the incoming call route for Incoming Number “7329450229” is
illustrated. The Line Group Id is 7, matching the Incoming Group field configured in the SIP
URI tab for the SIP Line to Verizon Business in Section 3.4.
Select the Destinations tab. From the Destination drop-down, select the extension to receive
the call when a PSTN user dials 7329450229. As shown in Table 1, 7329450229 is the DID
number associated with IP Office user extension 30026.
Incoming Call Routes for other direct mappings of DID numbers to IP Office users listed in
Table 1 are omitted here, but can be configured in the same fashion.
In the screen shown below, the incoming call route for Incoming Number “7329450236” is
illustrated. The Line Group Id is 7, matching the Incoming Group field configured in the SIP
URI tab for the SIP Line to Verizon Business in Section 3.4.
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Select the Destinations tab. From the Destination drop-down, select the extension to receive
the call when a PSTN user dials 7329450236. In this case, the destination is the hunt group “200
Main” whose configuration is shown in Section 3.5.3.
In the screen shown below, the incoming call route for Incoming Number “7329450237” is
illustrated. The Line Group Id is 7, matching the Incoming Group field configured in the SIP
URI tab for the SIP Line to Verizon Business in Section 3.4.
Select the Destinations tab. From the Destination drop-down, select the extension to receive
the call when a PSTN user dials 7329450237. In this case, the destination is the hunt group
“30200 IP Stations Seq” whose configuration is shown in Section 3.5.3.
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When configuring an Incoming Call Route, the Destination field can be manually configured
with a number such as a short code, or certain keywords available from the drop-down list. For
example, the following Destinations tab for an incoming call route contains the Destination
“*17” entered manually. The dial string “*17” is the short code for “Voicemail Collect”, as
shown in Section 3.6. An incoming call to 732-945-0234 will be delivered directly to voice
mail, allowing the caller to log-in to voicemail and access messages.
Similar, the following Destinations tab for an incoming call route contains the Destination
“*99” entered manually. The dial string “*99” is the short code for accessing the “Attendant”
application on Voicemail Pro, as shown in Section 3.6. An incoming call to 732-945-0228 will
be delivered directly to the Voicemail Pro “Attendant” application, which will allow the caller to
be prompted with announcements, navigate via DTMF, and transfer to IP Office users. The
configuration of the “Attendant” application on Voicemail Pro is outside the intended scope of
these Application Notes.
3.8. ARS and Alternate Routing
While detailed coverage of ARS is beyond the scope of these Application Notes, this section
includes basic ARS screen illustrations and considerations. ARS is illustrated here mainly to
illustrate alternate routing should the SIP Line be out of service or temporarily not responding.
Optionally, Automatic Route Selection (ARS) can be used rather than the simple “9N;” short
code approach documented in Section 3.6. With ARS, secondary dial tone can be provided after
the access code, time-based routing criteria can be introduced, and alternate routing can be
specified so that a call can re-route automatically if the primary route or outgoing line group is
not available. Although not shown in this section, ARS also facilitates more specific dialed
telephone number matching, enabling immediate routing and alternate treatment for different
types of numbers following the access code. For example, if all 1+10 digit calls following an
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access code should use the SIP Line preferentially, but other local or service numbers following
the access code should prefer a different outgoing line group, ARS can be used to distinguish the
call behaviors.
To add a new ARS route, right-click ARS in the Navigation pane, and select New. To view or
edit an existing ARS route, select ARS in the Navigation pane, and select the appropriate route
name in the Group pane.
The following screen shows an example ARS configuration for the route named “Main”. The In
Service parameter refers to the ARS form itself, not the Line Groups that may be referenced in
the form. If the In Service box is un-checked, calls are routed to the ARS route name specified
in the Out of Service Route parameter. IP Office short codes may also be defined to allow an
ARS route to be disabled or enabled from a telephone. The configurable provisioning of an Out
of Service Route, and the means to manually activate the Out of Service Route can be helpful for
scheduled maintenance or other known service-affecting events for the primary route.
Assuming the primary route is in-service, the number passed from the short code used to access
ARS (e.g., 7N in Section 3.6) can be further analyzed to direct the call to a specific Line Group
ID. Per the example screen above, if the user dialed 7-1-908-848-5704, the call would be
directed to Line Group 18, another SIP Line that exists in the configuration that is not described
in these Application Notes. If Line Group 18 cannot be used, the call can automatically route to
the route name configured in the Additional Route parameter in the lower right of the screen.
Since alternate routing can be considered a privilege not available to all callers, IP Office can
control access to the alternate route by comparing the calling user’s priority to the value in the
Alternate Route Priority Level field.
The following screen shows an example ARS configuration for the route named “backup”, ARS
Route ID 51. Continuing the example, if the user dialed 7-1-908-848-5704, and the call could
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not be routed via the primary route “50: Main” described above, the call will be delivered to this
“backup” route. Per the configuration shown below, the call will be delivered to Line Group 8,
using the SIP Line to Verizon Business IP Trunk service described in these Application Notes.
The configuration of the Code, Telephone Number, Feature, and Line Group ID for an ARS
route is similar to the configuration already shown for short codes in Section 3.6. In this case,
the originally dialed number (sans the short code 7) is delivered in the Request URI along with
the Verizon FQDN (i.e., the contents of the INVITE sent to Verizon are the same as the 9-1-908848-5704 short code 9 approach from Section 3.6). Additional codes (e.g., 411, 0+10, etc.) can
be added to the ARS route by pressing the Add… button to the right of the list of previously
configured codes.
In the testing associated with the configuration, calls were successfully delivered to SIP Line 8
via both the primary ARS route “50: Main” (via changes to “50: Main”) as well as the backup
ARS route shown above. If a primary route experiences a network outage such that no response
is received to an outbound INVITE, IP Office successfully routes the call via the backup route.
The user receives an audible tone when the re-routing occurs and may briefly see “Waiting for
Line” on the display. The testing verified that the INVITE was sent to the primary route, and
the call re-routed upon timeout. The call was made right after a failure of the primary route was
induced, so IP Office had not yet marked the SIP Line out of service as a result of no response to
SIP OPTIONS. Testing also verified that calls can be delivered to Verizon via the alternate route
when the primary route was manually marked out-of-service, or known to be out-of-service due
to prior failure of SIP OPTIONS.
3.9. Privacy / Anonymous Calls
As described in Section 3.6, an IP Office user whose calling line identification is not typically
withheld from the network can request privacy in the sample configuration by dialing the short
code 8 to access the SIP Line. The Avaya 1600-Series IP Telephones can also request privacy
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for a specific call, without dialing a unique short code, using Features  Call Settings 
Withhold Number. Specific users may be configured to always withhold calling line
identification by checking the Anonymous field in the SIP tab for the user (Section 3.5).
To configure IP Office to include the caller’s DID number in the P-Asserted-Identity SIP header,
required by Verizon Business to admit an otherwise anonymous caller to the network, the
following procedure may be used.
From the Navigation pane, select User. From the Group pane, scroll down past the configured
users and select the user named NoUser. From the NoUser Details pane, select the tab Source
Numbers. Press the Add… button to the right of the list of any previously configured Source
Numbers. In the Source Number field shown below, type SIP_USE_PAI_FOR_PRIVACY.
Click OK.
The source number SIP_USE_PAI_FOR_PRIVACY should now appear in the list of Source
Numbers as shown below.
3.10. SIP Options Frequency
In Section 3.4, the SIP Line to Verizon Business is shown with the Check OOS box checked. In
the sample configuration, IP Office periodically checks the health of the SIP Line by sending a
SIP OPTIONS message. If there is no response, IP Office can mark the trunk out of service.
Although ARS as shown in Section 3.8 can include alternate routes to complete calls even if the
far-end is not responding, IP Office must wait for the outbound INVITE to timeout before route
advance. Once the SIP OPTIONS maintenance recognizes that the SIP Line is out-of-service,
new calls will no longer be delayed before route advance. Also, once the problem with the SIP
Line is resolved, the SIP OPTIONS maintenance will automatically bring the link back to the inservice state.
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If a customer wishes to control how often SIP OPTIONS messages are sent by IP Office, a
NoUser Source Number can be configured as follows. This configuration complements the
configuration presented in Section 3.3.2 and Section 3.4
From the Navigation pane, select User. From the Group pane, scroll down past the configured
users and select the user named NoUser. From the NoUser Details pane, select the tab Source
Numbers. Press the Add… button to the right of the list of any previously configured Source
Numbers. In the Source Number field shown below, type SIP_OPTIONS_PERIOD=X. X is
a value (in minutes) representing a longer time than the interval configured (in seconds) in the
Binding Refresh Interval. In the sample configuration, the value used for X was 2 minutes.
Click OK.
The source number SIP_OPTIONS_PERIOD=2 should now appear in the list of Source
Numbers as shown below.
With this configuration, Binding Refresh Intervals of 30 seconds and 60 seconds were tested
successfully. That is, IP Office sourced SIP OPTIONS every 30 or 60 seconds, depending on the
value configured in the Binding Refresh Interval, since the Binding Refresh Interval was less
than the value configured via the SIP_OPTIONS_PERIOD source number.
3.11. Saving Configuration Changes to IP Office
When desired, send the configuration changes made in IP Office Manager to the IP Office
server, to cause the changes to take effect. Click the “floppy disk” icon that is the third icon
from the left (i.e., common “save” icon with mouse-over help “Save Configuration File”). Click
Yes to validate the configuration, if prompted.
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Once the configuration is validated, a screen similar to the following will appear, with either
“Merge” or “Immediate” selected, based on the nature of the configuration changes made since
the last save. Note that clicking OK may cause a service disruption. Click OK if desired.
4. Verizon Business Configuration
Information regarding Verizon Business IP Trunk service offer can be found by contacting a
Verizon Business sales representative, or by visiting
http://www.verizonbusiness.com/us/products/voip/trunking/.
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The reference configuration described in these Application Notes was located in the Avaya
Solutions and Interoperability Lab. The Verizon Business IP trunk service was accessed via a
Verizon Private IP (PIP) T1 connection. Verizon Business provided the necessary service
provisioning.
The following Fully Qualified Domain Names (FQDNs) were provided by Verizon for the
reference configuration.
CPE (Avaya)
adevc.avaya.globalipcom.com
Verizon Network
pcelban0001.avayalincroft.globalipcom.com
For service provisioning, Verizon will require the customer IP address used to reach the Avaya
IP Office server. Verizon provided the following information for the compliance testing: the IP
address and port used by the Verizon SIP SBC, and the Direct Inward Dialed (DID) numbers
shown in Figure 1 and Table 1. This information was used to complete the Avaya IP Office
configuration shown in Section 3.
5. Verifications
The Avaya IP Office location was connected to the Verizon Business IP Trunk Service, as
depicted in Figure 1. Avaya IP Office was configured to use the commercially available SIP
Trunking solution provided by the Verizon Business IP Trunk SIP Trunk Service. This allowed
Avaya IP Office users to make calls to the PSTN and receive calls from the PSTN via the
Verizon Business IP Trunk SIP Trunk Service.
5.1. Verification Summary
This section summarizes the verification testing associated with these Application Notes.
Successful SIP trunk interoperability compliance testing included the following:





Incoming calls from the PSTN were routed to the DID numbers assigned by Verizon
Business to the Avaya IP Office location. These incoming PSTN calls arrived via the SIP
Line and were answered by Avaya H.323 telephones, Avaya digital telephones, analog
telephones, Avaya IP Office Softphone, and Avaya IP Office Voicemail Pro voicemail
and auto-attendant applications. The display of caller ID on display-equipped Avaya IP
Office telephones was verified.
Incoming calls answered by members of collective and sequential Hunt Groups were
verified.
Outgoing calls from the Avaya IP Office location to the PSTN were routed via the SIP
Line to Verizon Business. These outgoing PSTN calls were originated from Avaya
H.323 telephones, Avaya digital telephones, analog endpoints, and Avaya IP Office
Softphone. The display of caller ID on display-equipped PSTN telephones was verified.
Proper disconnect when the caller abandons a call before answer for both inbound and
outbound calls.
Proper disconnect when the IP Office party or the PSTN party hang-up an active call
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
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




Proper busy tone heard when an IP Office user calls a busy PSTN user, or a PSTN user
calls a busy IP Office user (i.e., if no redirection is configured for user busy conditions)
Various outbound PSTN call types were tested including long distance, international,
toll-free, operator assisted, and directory assistance calls.
Requests for privacy (i.e., caller anonymity) for IP Office outbound calls to the PSTN
were verified. That is, when privacy is requested by IP Office, outbound PSTN calls can
be successfully completed while withholding the caller ID from the displays of displayequipped PSTN telephones.
Privacy requests for inbound calls from the PSTN to IP Office users were verified. That
is, when privacy is requested by a PSTN caller, the inbound PSTN call can be
successfully completed to an IP Office user while presenting an “anonymous”
display to the IP Office user.
SIP OPTIONS monitoring of the health of the SIP trunk was verified. Both Verizon
Business and IP Office can monitor health using SIP OPTIONS. The Avaya IP Office
configurable control of SIP OPTIONS timing was exercised successfully.
IP Office outbound calls were placed with simple short codes as well as using ARS.
Using ARS, the ability of IP Office to route-advance to an alternate route when the
primary SIP line is not responding was exercised. The Line Group associated with the
Verizon Business SIP Line can be the primary line group chosen for a call, or an alternate
line group selected upon failure of a primary line.
Incoming and outgoing calls using the G.729(a) and G.711 ULAW codecs.
DTMF transmission using RFC 2833 with successful voice mail navigation for G.729a
and G.711MU for incoming and outgoing calls. Successful navigation of a simple autoattendant application configured on IP Office Voicemail Pro.
The “callback” feature of Avaya Voicemail Pro was tested successfully. When a
message was left for a voice mail subscriber with “callback” configured, an outbound call
was placed to the subscriber’s configured mobile telephone via the SIP Line to Verizon
Business. Upon answer, Voicemail Pro announces the call and prompts the user to enter
the “#” key to accept the call. The user has the opportunity to navigate the voicemail TUI
via DTMF (e.g., to listen to the voice message that stimulated the callback).
Inbound and outbound long holding time call stability
Telephony features such as call waiting, hold, transfer, and conference.
Call Forwarding to PSTN destinations (see Section 1.1)
Mobile twinning to a mobile phone, presenting true calling party information to the
mobile phone. Outbound mobile call control was also verified successfully (e.g., using
DTMF on a twinned call to place new calls and create a conference via a mobile phone).
Proper DiffServ markings for IP Office SIP signaling and RTP media
5.2. System Status Application
The System Status application can be used to monitor or troubleshoot IP Office. The System
Status application can typically be accessed from Start  Programs  IP Office  System
Status. See reference [IPO-SYSSTAT] for more information.
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The Navigation pane allows various views into the system status. The following screen shows
an example Extension Summary report, accessible by selecting Extensions from the navigation
pane as shown below.
At the time this summary report was captured, the IP Office Softphone was logged in as user
30026, causing the Avaya 1616 telephone that had been logged in as 30026 to enter “NoUser”
mode (as expected). This can be observed in the last two rows below. In the final row, note that
the Telephone Type for the Avaya IP Office Softphone logged in as Current User Extension
30026 is shown as “SIP Softphone”, with an arbitrary Home Extension Number.
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The following screen shows an example Status tab for Trunks  Line 7, the SIP Line to
Verizon. In this case, the screen was captured while there was an incoming call active to
extension 30025, an Avaya 1616 IP Telephone. As can be observed, the call used Codec G.729a
and Connection Type RTP relay.
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The following screen shows an example Extension Status tab for Extensions  30025. In this
case, the screen was captured while there was an active incoming PSTN SIP Trunk call from
PSTN user 908-848-5704 to extension 30025. Again, it can be observed that the call used
G.729a and RTP relay. The Remote RTP Address (172.30.209.132) is in the Verizon network
(i.e., signaled by Verizon in the SIP SDP).
5.3. System Monitor Application
The System Monitor application can also be used to monitor or troubleshoot. The System
Monitor application can typically be accessed from Start  Programs  IP Office 
Monitor. See reference [IPO-MON] for more information.
The application allows the monitored information to be customized. To customize, select the
button shown within the red rectangle below, or Filters  Trace Options.
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The following screen shows the SIP tab, allowing configuration of SIP monitoring. In this
example, all SIP messages will appear in the trace with the color blue. To customize the color,
right-click on SIP Rx or SIP Tx and select the desired color.
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The following shows a portion of an example SIP trace for an incoming call. This example
window shows the inbound SIP INVITE from Verizon, and the 100 Trying returned by IP
Office. In this case, the call was from PSTN user 908-848-5704 to the Verizon DID 732-9450231.
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The following shows a portion of an example trace for an outbound call. This example window
shows the outbound SIP INVITE to Verizon, and the 100 Trying returned by Verizon. In this
case, IP Office user 201 is calling PSTN user 1-908-848-5704. IP Office user 201 corresponds
to Verizon DID 732-945-0231, which can be observed in the From header.
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5.4. IP Office Softphone Application
The following screen shows the IP Office Softphone Login screen. In this case, the Profile “IP
Office: Default” is chosen. The Login server is the IP Address of IP Office as configured in
Section 3.1. In the Username and Password fields, enter the values configured for the Name
and Password fields in the IP Office User tab, as shown in Section 3.5.
The following screen shows the IP Office Softphone Login screen. In this case, the Profile “IP
Office: Low Bandwidth” is chosen. As noted in Section 1.1, if use of G.729a is desired for both
inbound and outbound calls using the SIP Line configured in Section 3.4, the “IP Office: Low
Bandwidth” option can be chosen when logging in to the IP Office Softphone.
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The following screen capture shows the IP Office Softphone interface on an outbound PSTN
call. In this case, the IP Office Softphone user dialed the PSTN user 908-848-5704 via the “9N;”
short code illustrated in Section 3.6.
The following screen capture shows a portion of the IP Office Softphone interface as an inbound
PSTN call is ringing the user. In this case, a mobile phone (732-687-0755) on the PSTN dialed
732-945-0229, and the Incoming Call Route illustrated in Section 3.7 sent the call to user 30026.
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The IP Softphone user is currently logged in as 30026 and receives the call. If the Answer
button is pressed, the call is answered successfully. If the Decline button is pressed, the call is
redirected to the user’s voice mail greeting, since this user has voice mail. The words
“WIRELESS CALLER” appearing on the display were included in the Display Information in
the From header of the inbound INVITE from Verizon.
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The following screen capture shows a portion of the IP Office Softphone interface as an inbound
“private / anonymous” PSTN call is alerting the user. In this case, the same mobile phone (732687-0755) on the PSTN dialed *67-732-945-0229, and the Incoming Call Route illustrated in
Section 3.7 sent the call to user 30026. For this mobile telephone, dialing *67 before the
destination number is a request for privacy (i.e., a request to prevent the presentation of the
calling party identity to the called party). The IP Softphone user is currently logged in as 30026
and receives the call. Observe the anonymous display.
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The following screen capture shows a portion of the IP Office Softphone interface after an
inbound PSTN call has been answered. In this case, a PSTN phone (908-848-5704) dialed 732945-0229, and the Incoming Call Route illustrated in Section 3.7 sent the call to user 30026. The
IP Softphone user is currently logged in as 30026 answers the call. The words “AVAYA
ALPHA” appearing on the display were included in the From header of the inbound INVITE
from Verizon.
The IP Office Softphone includes integrated diagnostics capabilities. For example, the following
screen can be viewed by selecting Help  Troubleshooting and selecting the Audio tab.
The following screen can be viewed by selecting the Diagnostics tab from the Troubleshooting
screen shown above. The Information and Components drop-down menus allow customization
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of the level and type of troubleshooting information to be gathered. The diagnostic information
can be written to a file using the Save a log file of my system and connection information
check box, and the log files can be accessed via the Open Folder button.
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5.5. Mobile Twinning Application
As shown in Section 3.5, user 201 is configured for Mobile Twinning. When an incoming call
rings to user 201, the call is “twinned out” to the user’s mobile phone, configured as 9-1-732687-0755. The Mobile Delay Time on the Mobility tab for the user was set to 2 seconds.
Therefore, the outbound INVITE to Verizon will be sent after 2 seconds of ringing the IP Office
user. If the IP Office user answers before the Mobile Delay Time, the call will not be twinned
out to Verizon. If the IP Office user answers after the INVITE message has been sent to
Verizon, and before the mobile device answers, IP Office will cancel the outbound call.
The following portion of a filtered Wireshark trace illustrates the behavior for an incoming
PSTN call from 908-848-5704 to 732-945-0231 (user 201) that is answered by the IP Office user
after the INVITE has been sent to Verizon. In frame 12, the INVITE arrives from Verizon for
DID 7329450231, whose IP Office Incoming Call Route maps to destination 201. In frame 15
two seconds later, IP Office sends an INVITE to Verizon to the PSTN number configured as the
mobile telephone for user 201. In frame 75, IP Office sends a CANCEL because the local IP
Office user has answered. In frame 77, the IP Office sends the 200 OK answering the incoming
call.
The following portion of a filtered Wireshark trace illustrates the behavior for an incoming
PSTN call from 908-848-5704 to user 201 that is answered by the mobile telephone. In frame 4,
the INVITE arrives from Verizon for DID 7329450231. In frame 7 two seconds later, IP Office
sends an INVITE to Verizon to the PSTN number configured as the mobile telephone for user
201. In frame 385, Verizon sends a 200 OK when the mobile telephone user answers the call. In
frame 388, IP Office sends the 200 OK answering the incoming call.
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This same trace can be further analyzed to reveal the Diversion information included in the
outbound INVITE to Verizon for the twinned outbound call leg. The following screen shows
frame 7 expanded to show the SIP information. Note that the From header contains the original
calling party information, allowing the mobile telephone receiving the twinned call to see the
true caller’s identity. As can be observed in the last line below, the Diversion header inserted by
IP Office includes the DID associated with the IP Office twinning user, 7329450231. This
allows Verizon to admit the outbound twinned call to the network.
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5.6. Outbound Anonymous / Private Calls
As shown in Section 3.6, a user may dial using the short code 8 to request privacy. In the
filtered Wireshark trace shown below, user 201 dialed 8-1-908-848-5704. The INVITE sent by
IP Office in frame 7 is expanded. Observe that IP Office includes “Privacy: Id”, with the PAsserted-Identity (PAI) Header containing the DID of the user. Verizon completes the call to the
called PSTN user, and the called telephone does not see the caller id information. Section 3.4
and Section 3.9 show the configuration required to cause IP Office Release 6 to include the PAI
Header, as required by Verizon for an anonymous call.
5.7. SIP OPTIONS
IP Office can be configured to maintain a SIP Line using SIP OPTIONS. Sections 3.3, 3.4, and
3.10 illustrate the pertinent configuration. In the filtered Wireshark trace shown below, SIP
OPTIONS and 200 OK responses can be observed. In frame 37, IP Office sends OPTIONS to
Verizon, and Verizon responds with 200 OK in frame 40. Coincidentally, Verizon sent SIP
OPTIONS to IP Office at about the same time, with the OPTIONS from Verizon shown in frame
38 and the 200 OK from IP Office in frame 39. Frame 150 shows another OPTIONS sent by IP
Office, 60 seconds after the OPTIONS in frame 37. (At the time of this trace, the Binding
Refresh Time for LAN1 was set to 60 seconds.)
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5.8. DiffServ Markings by IP Office
As shown in Section 3.3, IP Office can mark the Differentiated Services Field in the IP Header to
configurable values for SIP signaling and RTP. For example, the following portion of a filtered
Wireshark trace shows the SIP INVITE message sent by IP Office. As can be observed, the
Differentiated Services Field in the IP Header is decoded as “Assured Forwarding 32”, as
configured in Section 3.3.2 for SIP signaling. Other values may be used.
The following portion of a filtered Wireshark trace shows an example RTP packet sent by IP
Office. As can be observed, the Differentiated Services Field in the IP Header is decoded as
“Expedited Forwarding”, as configured in Section 3.3.2 for RTP. Other values may be used. It
can also be observed that the IP Office UDP port for RTP (49152) is within the range configured
in Section 3.3.2. Note that the IP Office IP Address 1.1.1.2 is always the source for the RTP
from IP Office towards Verizon, regardless of whether the telephone device at the customer
location is analog, digital, H.323, or SIP.
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6. Support
6.1. Avaya
For technical support on the Avaya VoIP products described in these Application Notes visit
http://support.avaya.com
6.2. Verizon
For technical support on Verizon Business IP Trunk service offer, visit the online support site at
http://www.verizonbusiness.com/us/customer/
7. Conclusion
IP Office is a highly modular IP telephone system designed to meet the needs of home offices,
standalone businesses, and networked branch and head offices for small and medium enterprises.
These Application Notes demonstrated how IP Office can be successfully combined with a
Verizon Business IP Trunk SIP trunk service connection to create an end-to-end SIP Telephony
business solution. Through following the example configurations provided in this document,
customers using Avaya IP Office can now connect to the PSTN via a Verizon Business IP Trunk
SIP Trunk service connection, thus eliminating the costs of analog or digital trunk connections
previously required to access the PSTN. Utilizing this solution, IP Office customers can now
leverage the operational efficiencies and cost savings associated with SIP trunking while gaining
the advanced technical features provided through the marriage of best of breed technologies from
Avaya and Verizon.
8. References
This section references documentation relevant to these Application Notes. In general, Avaya
product documentation is available at http://support.avaya.com
[IPO-INSTALL] IP Office 6.0 Installation Manual, Issue 21f, March 1 2010
Document Number 15-601042
http://support.avaya.com/css/P8/documents/100073460
[IPO-MGR] IP Office Release 6.0 Manager 8.0, Issue 24h, February 20, 2010
Document Number 15-601011
http://support.avaya.com/elmodocs2/ip_office/R4.2/Newissuesept08/eng/manager_en.pdf
[IPO-SYSSTAT] IP Office Release 6.0 System Status Application, Issue 05a, February 12, 2010
Document Number 15-601758
http://support.avaya.com/css/P8/documents/100073300
[IPO-VMPRO] IP Office Release 6.0 Voicemail Pro, Issue 22b, January 16, 2010
Document Number 15-601063
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http://support.avaya.com/css/P8/documents/100073435
[IPO-MON] IP Office System Monitor, Issue 02b, November 28, 2008
Document Number 15-601019
http://support.avaya.com/css/P8/documents/100073350
Additional IP Office documentation can be found at:
http://marketingtools.avaya.com/knowledgebase/
Avaya IP Office Video Softphone documentation can be found here:
http://marketingtools.avaya.com/knowledgebase/businesspartner/ipoffice/mergedProjects/softph
oneuser/
[CM-VZIPT] Application Notes for Avaya Aura™ Communication Manager 5.2, Avaya
Aura™ Session Manager 1.1, and Acme Packet 3800 Net-Net Session Director integration with
Verizon Business IP Trunk SIP trunk service offer – Issue 1.3
https://devconnect.avaya.com/public/download/dyn/AvayaSM_VzB_IPT.pdf
[RFC-3261] RFC 3261 SIP: Session Initiation Protocol http://www.ietf.org/rfc/rfc3261.txt
[RFC-2833] RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
http://www.ietf.org/rfc/rfc2833.txt
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©2010
Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
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