Application Notes for Configuring Cbeyond SIP Trunking with Avaya

Application Notes for Configuring Cbeyond SIP Trunking with Avaya
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Cbeyond SIP Trunking
with Avaya IP Office 7.0 - Issue 1.0
Abstract
These Application Notes describe the procedures for configuring Session Initiation Protocol
(SIP) Trunking between Cbeyond’s BeyondVoice with sipConnect service and Avaya IP Office
7.0.
The BeyondVoice with sipConnect service provides PSTN access via a SIP trunk between small
business sites and the Cbeyond network as an alternative to legacy analog or digital trunks. This
approach generally results in lower cost for the business.
Cbeyond is a member of the Avaya DevConnect Service Provider program. Information in these
Application Notes has been obtained through DevConnect compliance testing and additional
technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution
and Interoperability Test Lab.
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1. Introduction
These Application Notes describe the procedures for configuring Session Initiation Protocol (SIP)
Trunking between Cbeyond’s BeyondVoice with sipConnect service and Avaya IP Office 7.0. In
the sample configuration, the Avaya IP Office solution consists of an Avaya IP Office 500v2
Release 7.0, Avaya Voicemail Pro, SIP-based Avaya IP Office Softphone, and Avaya H.323, digital
and analog endpoints.
The Cbeyond SIP Trunking service referenced within these Application Notes is designed for small
business customers. The service enables local and long distance PSTN calling via standards-based
SIP trunks as an alternative to legacy analog or digital trunks, without the need for additional TDM
gateways and the associated maintenance costs.
The Cbeyond SIP Trunking service uses Digest Authentication for SIP trunk registration as well as
for outbound calls from the small business site, using challenge-response authentication for each call
to the Cbeyond network based on a configured user name and password (provided by Cbeyond and
configured in IP Office). This call authentication scheme, as specified in SIP RFC 3261, provides
security and integrity protection for SIP signaling.
2. General Test Approach and Test Results
The general test approach was to configure a simulated small business site using Avaya IP Office to
connect to Cbeyond SIP Trunking service. This configuration (shown in Figure 1) was used to
exercise the features and functionality tests listed in Section 2.1.
2.1. Interoperability Compliance Testing
A simulated small business site with Avaya IP Office was connected to Cbeyond SIP Trunking
service. To verify SIP trunking interoperability, the following features and functionality were
exercised during the interoperability compliance test:
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Response to SIP OPTIONS queries.
Incoming PSTN calls to various phone types at the small business site. Phone types included
H.323, digital and analog telephones. All inbound PSTN calls were routed to the small
business site across the SIP trunk from the service provider.
Outgoing calls to the PSTN from various phone types. Phone types included H.323, digital
and analog telephones. All outbound PSTN calls were routed from the small business site
across the SIP trunk to the service provider.
Inbound and outbound PSTN calls to/from the Avaya IP Office Softphone.
Inbound and outbound long-duration call stability.
Inbound and outbound long hold time call stability.
Various call types including: local, long distance, international, outbound toll-free, operator
service and directory assistance.
G.711MU codec.
Caller number/ID presentation.
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Privacy requests (i.e., caller anonymity) and Caller ID restriction for inbound and outbound
calls.
DTMF transmission using RFC 2833.
Voicemail navigation for inbound and outbound calls.
Telephony features such as hold and resume, transfer, and conference.
Use of SIP REFER for call redirection to the PSTN.
Off-net call forwarding and transfer.
Twinning to mobile phones on inbound calls.
2.2. Test Results
Cbeyond SIP Trunking passed compliance testing.
Items not supported or not tested included the following:
 Inbound toll-free and outbound emergency calls (911) are supported but were not tested as
part of the compliance test.
 Faxing between the local site and PSTN were not tested as part of the compliance test since
Cbeyond currently does not support T.38 FoIP (Fax over IP) for its SIP Trunking service.
2.3. Observations and Limitations
Interoperability testing of Cbeyond SIP Trunking was completed with successful results for all test
cases with the exception of the observations/limitations described below.
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OPTIONS from Cbeyond – Cbeyond configured its SIP Trunking circuit not to send
OPTIONS to the small business site, but it responded to OPTIONS from the small business
site properly.
Codec Mismatch – Cbeyond SIP Trunking service returned “480 Temporarily Unavailable”
for outbound calls from IP Office to the PSTN instead of a more appropriate status code like
“488 Not Acceptable Here”. For inbound calls from the PSTN to IP Office, the IP Office
treated this situation properly, returning “488 Not Acceptable Here”.
Call Display on PSTN Phone – Call display was not properly updated on the PSTN phone
involved in a call transfer. After the call transfer was completed, the PSTN phone did not
display the actual connected party but instead showed the party that initiated the transfer.
SIP signaling traces showed that IP Office did not send an UPDATE message to the network
to update the call display of the PSTN phone in the call transfer. This problem was reported
to Avaya IP Office for further investigation.
Operator-Assisted Call – Operator assisted calls (0 + 10-digits) did not reach a live
operator or an operator voice-menu service, but was routed to the destination directly.
Offnet Call Transfer / Forward – When an inbound call was transferred or forwarded back
out to the PSTN, CBeyond responded to the REFER from IP Office with "401
Unauthorized" instead of "202 Accepted". However, user experience of the call scenarios
was not negatively affected.
Media Deadlock on Off-Net Call Redirection – When an inbound call was forwarded back
to the PSTN or twinned to a mobile phone, there was no audio either way on the established
re-directed call. This problem was caused by media deadlock where Cbeyond expected an
IP Office endpoint to send RTP packets first even though there was no IP Office media
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endpoint involved in this call scenario since the call had been re-directed back to the PSTN.
The compliance test worked around this problem by enabling the RTP Keepalives
configuration in IP Office (see Section 5.1) to send bogus RTP packets toward the service
provider at the start of the call for breaking the media deadlock. This problem was reported
to Cbeyond for proper network resolution.
Early Media – Early Media on inbound calls (responding to INVITEs with 18X with SDP)
is not supported by IP Office Release 7.0.
Hold/Resume – There is no SIP signaling to the network when an active call was placed on
hold or resumed from hold at an IP Office extension. This is by design with IP Office.
Session Refresh – It was observed during the compliance test that no session-refresh reINVITE or UPDATE messages were issued from IP Office towards the network during
long-duration calls. This was normal since there was no “timer” specification in the
Supported header in SIP messages from Cbeyond. Cbeyond issued session-refresh
UPDATE messages towards IP Office properly.
For technical support on the Avaya products described in these Application Notes visit
http://support.avaya.com.
For technical support on Cbeyond’s BeyondVoice with sipConnect service, contact Cbeyond at
http://www.cbeyond.net/small-business-solutions/voice-broadband/beyondvoice-with-sipconnect/.
3. Reference Configuration
Figure 1 below illustrates the test configuration. The test configuration shows a small business site
connected to Cbeyond SIP Trunking service through the public IP network.
Located at the small business site is an Avaya IP Office 500v2 with the COMBO6210/ATM4
expansion card which provides connections for 6 digital stations, 2 analog stations, 4 analog trunks
to the PSTN as well as 10-channel VCM (Voice Compression Module) for supporting VoIP codecs.
The LAN port of Avaya IP Office is connected to the LAN of the small business site while the WAN
port is connected to the public IP network. Endpoints include an Avaya 1600 Series IP Telephone
(with H.323 firmware), an Avaya 4600 Series IP Telephone (with H.323 firmware), an Avaya 9600
Series IP Telephone (with H.323 firmware), Avaya 5410 and 6408D Digital Telephones, an Avaya
6210 Analog Telephone and a SIP-Based Avaya IP Office Softphone. The site also has a Windows
2003 Server running Avaya Voicemail Pro for providing voice messaging service to the Avaya IP
Office users. A separate Windows XP PC runs Avaya IP Office Manager to configure and
administer the Avaya IP Office.
Mobility Twinning is configured for some of the Avaya IP Office users so that calls to these user’s
desk phones will also ring and can be answered at the configured mobile phones.
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Figure 1: Test Configuration for Avaya IP Office with Cbeyond SIP Trunking Service
For security purposes, the real public IP addresses and PSTN routable phone numbers used in the
compliance test are masked in these Application Notes.
Note that Cbeyond assigns a prescribed domain name identical for both the small business site and
the network service.
For the purposes of the compliance test, Avaya IP Office users dialed a short code of 9 + N digits to
send digits across the SIP trunk to Cbeyond. The short code of 9 was stripped off by Avaya IP Office
but the remaining N digits were sent unaltered to Cbeyond. For calls within the North American
Numbering Plan (NANP), the user would dial 11 (1 + 10) digits. Thus for these NANP calls, Avaya
IP Office would send 11 digits in the Request URI and the To header of an outbound SIP INVITE
message. It was configured to send 10 digits in the From header. For inbound calls, Cbeyond SIP
Trunking sent 10 digits in the Request URI and the To header of inbound SIP INVITE messages.
In an actual customer configuration, the small business site may also include additional network
components between the service provider and Avaya IP Office such as a session border controller or
data firewall. A complete discussion of the configuration of these devices is beyond the scope of
these Application Notes. However, it should be noted that SIP and RTP traffic between the service
provider and Avaya IP Office must be allowed to pass through these devices.
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4. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration provided:
Avaya Telephony Components
Equipment
Avaya IP Office 500v2
Avaya IP Office COMBO6210/ATM4
Module
Avaya Voicemail Pro
Avaya IP Office Manager
Avaya 1616 IP Telephone (H.323)
Avaya 4621SW IP Telephone (H.323)
Avaya 9620 IP Telephone (H.323)
Avaya Digitial Telephones (5410)
Avaya Digitial Telephones (6408D)
Avaya 6210 Analog Telephone
Avaya IP Office Softphone (SIP)
Release
7.0 (232702)
7.0 (232702)
7.0.23
9.0 (232702)
Avaya one-X Deskphone Value Edition 1.2.2
2.9.1
Avaya one-X Deskphone Edition 3.1
N/A
N/A
N/A
3.1.2.17 59616
5. Configure IP Office
This section describes the Avaya IP Office configuration necessary to support connectivity to
Cbeyond SIP Trunking service. Avaya IP Office is configured through the Avaya IP Office
Manager PC application. From a PC running the Avaya IP Office Manager application, select Start
 Programs  IP Office  Manager to launch the application. Navigate to File  Open
Configuration, select the proper Avaya IP Office system from the pop-up window, and log in with
the appropriate credentials. A management window will appear similar to the one shown in the next
section. 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. These panes will be
referenced throughout the Avaya IP Office configuration. Proper licensing as well as standard
feature configurations that are not directly related to the interface with the service provider (such as
LAN interface to the local site and IP Office Softphone support) is assumed to already be in place.
5.1. LAN2 Settings
In the sample configuration, the MAC address 00E007055793 was used as the system name and the
WAN port was used to connect the Avaya IP Office to the public network. The LAN2 settings
correspond to the WAN port on Avaya IP Office. To access the LAN2 settings, first navigate to
System (1)  00E007055793 in the Navigation and Group Panes and then navigate to the LAN2
LAN Settings tab in the Details Pane. Set the IP Address field to the IP address assigned to the
Avaya IP Office WAN port. Set the IP Mask field to the mask used on the public network. All
other parameters should be set according to customer requirements.
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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 Cbeyond. The RTP Port Number Range can
be customized to a specific range of receive ports for the RTP media. Based on this setting, Avaya
IP Office would request RTP media be sent to a UDP port in the configurable range for calls using
LAN2. Avaya IP Office can also be configured to mark the Differentiated Services Code Point
(DSCP) in the IP Header with specific values to support Quality of Services policies for both
signaling and media. The DSCP field is the value used for media and the SIG DSCP field is the
value used for signaling. The specific values used for the compliance test are shown in the example
below.
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Scroll down to the RTP Keepalives section. Select RTP for Scope; select Enabled for Initial
keepalives; enter 30 for Periodic timeout. These settings cause IP Office to send bogus RTP
packets toward the service provider at the start of the call to work around media deadlock in off-net
call forwarding or mobile twinning. See the bullet item Media Deadlock on Off-Net Call
Redirection in Section 2.3 for an explanation of this workaround.
All other parameters should be set according to customer requirements.
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On the Network Topology tab in the Details Pane, configure the following parameters:
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Select the Firewall/NAT Type from the pull-down menu that matches the network
configuration. No firewall or network address translation (NAT) device was used in the
compliance test as shown in Figure 1, so the parameter was set to Open Internet. With this
configuration, STUN will not be used.
Set Binding Refresh Time (seconds) to 60. This value is used as one input to determine the
frequency at which Avaya IP Office will send SIP OPTIONS messages to the service
provider. See Section 5.10 for complete details.
Set Public IP Address to the IP address of the Avaya IP Office WAN port.
All other parameters should be set according to customer requirements.
In the compliance test, the LAN1 interface was used to connect the Avaya IP Office to the local site
IP network. The LAN1 interface configuration is not directly relevant to the interface with Cbeyond
SIP Trunking service, and therefore is not described in these Application Notes.
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5.2. System Telephony Settings
Navigate to the Telephony  Telephony Tab in the Details Pane. Set the Automatic Codec
Preference for the default codec to be used for intra-site traffic. Choose the Companding Law
typical for the local site. For North America, ULAW is used. Uncheck the Inhibit Off-Switch
Forward/Transfer box to allow call forwarding and call transfer to the PSTN via the service
provider across the SIP trunk.
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5.3. Twinning Calling Party Settings
When using twinning, the calling party number displayed on the twinned phone is controlled by two
parameters. The first parameter is the Send original calling party information for Mobile
Twinning box on the SystemTwinning tab. The second parameter is the Send Caller ID
parameter on the SIP Line form (shown in Section 5.4).
If Send original calling party information for Mobile Twinning on the SystemTwinning tab is
checked, the setting of the second parameter is ignored and Avaya IP Office will send the following
in the SIP From Header:
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On calls from an internal extension to a twinned phone, Avaya IP Office will send the calling
party number of the originating extension.
On calls from the PSTN to a twinned phone, Avaya IP Office will send the calling party
number of the host phone associated with the twinned destination (instead of the number of
the originating caller).
The above behavior in Avaya IP Office Release 7 is the same as in Avaya IP Office Release 6 and
was tested and verified in the compliance test.
Avaya IP Office Release 7 also provides an alternative method of sending caller ID through SIP
Diversion headers (configured via unchecking Send original calling party information for Mobile
Twinning here and then selecting Diversion Header for the Send Caller ID parameter on the SIP
Line form in Section 5.4). This alternative configuration could provide more accurate caller ID
information if the service provider supports the SIP Diversion header.
For the compliance test, the Send original calling party information for Mobile Twinning box in
the SystemTwinning tab was checked which overrides any setting of the Send Caller ID
parameter on the SIP Line form.
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5.4. Administer SIP Line
A SIP line is needed to establish the SIP connection between Avaya IP Office and Cbeyond SIP
Trunking service. To create a SIP line, begin by navigating to Line in the left Navigation Pane, then
right-click in the Group Pane and select New  SIP Line. On the SIP Line tab in the Details Pane,
configure the parameters as shown below:
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Set ITSP Domain Name to the domain for Cbeyond’s BeyondVoice with sipConnect service
(prescribed by Cbeyond) so that IP Office uses this domain as the host portion of the SIP
URI in SIP headers such as the From header. In the compliance test, this assigned domain is
“sipconnect-fca-atl0.cbeyond.net”.
Set Send Caller ID to None. For the compliance test, this parameter was ignored since Send
original calling party information for Mobile Twinning is checked in Section 5.3.
Check the In Service box.
Check the Check OOS box. With this option selected, IP Office will use the SIP OPTIONS
method to periodically check the SIP Line.
Default values may be used for all other parameters.
The area of the screen entitled REFER Support is used to enable/disable SIP REFER for call redirection to the PSTN as in call transfer and forward scenarios. The default values of “Auto” for
Incoming and Outgoing effectively disable use of SIP REFER. To enable SIP REFER, select
“Always” from the drop-down menu for Incoming and Outgoing. In the compliance test, both
scenarios were successfully tested to transfer a call between a PSTN phone and a phone at the local
site, to a second PSTN phone.
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Select the Transport tab. The ITSP Proxy Address is set to the Cbeyond SIP Proxy IP Address
provided by Cbeyond. As shown in Figure 1, this IP Address is 207.236.222.111. 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 Cbeyond. The Use Network Topology Info parameter is set to LAN 2. This
associates the SIP Line with the parameters in the System  LAN2  Network Topology tab.
Other parameters retain default values in the screen below.
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A SIP Credentials entry must be created for Digest Authentication used by Cbeyond SIP trunking
service to authenticate calls from the small business site to the PSTN. To create a SIP Credentials
entry, first select the SIP Credentials tab. Click the Add button and the New Channel area will
appear at the bottom of the pane. 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. The entry was created with the
parameters shown below:
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Set User name and Authentication Name to the value provided by the service provider
(masked in the screen below).
Set Password to the value provided by the service provider.
Check the Registration required option. Cbeyond requires SIP trunk registration for Digest
Authentication.
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A SIP URI entry must be created to match each incoming number that Avaya IP Office will accept
on this line. Select the SIP URI tab and then click the Add button and the New Channel area will
appear at the bottom of the pane. To edit an existing entry, click an entry in the list at the top and
click the Edit… button. In the example screen below, a previously configured entry is edited. For
the compliance test, a single SIP URI entry was created that matched any DID number assigned to
an Avaya IP Office user. The entry was created with the parameters shown below:
 Set Local URI, Contact and Display Name to Internal Data. This setting allows calls on
this line when the SIP URI matches the number set in the SIP tab of any User, as shown in
Section 5.6.
 Set PAI to Internal Data. With this setting IP Office will populate the SIP P-AssertedIdentity header on outgoing calls with the data set in the SIP tab of the originating User, as
shown in Section 5.6.
 For Registration, select the account credentials previously configured on the line's SIP
Credentials tab.
 Associate this line with an incoming line group in the Incoming Group field. This line
group number will be used in defining incoming call routes for this line. Similarly, associate
the line to an outgoing line group using the Outgoing Group field. For the compliance test, a
new incoming and outgoing group 17 was defined that only contains this line (line 17).
 Set Max Calls per Channel to the number of simultaneous SIP calls that are allowed using
this SIP URI pattern.
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Select the VoIP tab to set the Voice over Internet Protocol parameters of the SIP line. Set the
parameters as shown below:
.
 The Compression Mode was configured using the Advanced button, allowing an explicit
ordered list of codecs to be specified. Checking the G.711 ULAW 64K codec causes Avaya
IP Office to include this codec, the only codec supported by the Cbeyond SIP Trunking
service, in the Session Description Protocol (SDP) offer.
 Set the DTMF Support field to RFC2833. This directs Avaya IP Office to send DTMF
tones using RTP event messages as defined in RFC2833.
 Select None for Fax Transport Support T.38 faxing is not currently supported by
Cbeyond. If G.711 pass-through faxing is supported by the service provider, G.711 should be
selected here (not tested in the compliance test).
 Check the Re-invite Supported box.
 Default values may be used for all other parameters.
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5.5. Short Code
Define a short code to route outbound traffic to the SIP line. To create a short code, select Short
Code in the left Navigation Pane, then right-click in the Group Pane and select New. On the Short
Code tab in the Details Pane, configure the parameters for the new short code to be created. The
screen below shows the details of the previously administered “9N;” short code used in the test
configuration.
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In the Code field, enter the dial string which will trigger this short code followed by a semicolon. In this case, 9N;. This short code will be invoked when the user dials 9 followed by
any number.
Set Feature to Dial. This is the action that the short code will perform.
Set Telephone Number to N”@sipconnect-fca.atl0.cbeyond.net:5060”. This field is used
to construct the Request URI and To headers in the outgoing SIP INVITE message. The
value N represents the number dialed by the user. The host part following the “@” is the
domain for Cbeyond’s SIP Trunking service.
Set the Line Group Id to the outgoing line group number defined on the SIP URI tab on the
SIP Line in Section 5.4. This short code will use this line group when placing the outbound
call.
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5.6. User
Configure the SIP parameters for each user that will be placing and receiving calls via the SIP line
defined in Section 5.4. To configure these settings, first select User in the left Navigation Pane, then
select the name of the user to be modified in the center Group Pane. Next, select the SIP tab in the
Details Pane. In the example below, the name of the user is “Allan”.
The values entered for the SIP Name and Contact fields are used as the user part of the SIP URI in
the From header for outgoing SIP trunk calls. They also allow matching of the SIP URI for incoming
calls without having to enter this number as an explicit SIP URI for the SIP line (Section 5.4). The
example below shows the settings for user “Allan”. The SIP Name and Contact are set to one of
the DID numbers assigned to the subscriber from Cbeyond. 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.
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One of the H.323 IP Phones at the business site uses the Mobile Twinning feature. The following
screen shows the Mobility tab for User “Allan”. 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, including the preceding 9 for the short code 9N; configured in
Section 5.5. Other options can be set according to customer requirements.
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5.7. Incoming Call Route
An incoming call route maps an inbound DID number on a specific line to an internal extension.
This procedure should be repeated for each DID number provided by the service provider. To create
an incoming call route, select Incoming Call Route in the left Navigation Pane, then right-click in
the center Group Pane and select New. On the Standard tab of the Details Pane, enter the
parameters as shown below:
 Set the Bearer Capacity to Any Voice.
 Set the Line Group Id to the incoming line group of the SIP line defined in Section 5.4.
 Set the Incoming Number to the incoming number on which this route should match.
 Default values can be used for all other fields.
On the Destinations tab, select the destination extension from the pull-down menu of the
Destination field. In this example, incoming calls to 416-555-0397 on line 17 are routed to user
Allan at extension 251.
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5.8. Privacy/Anonymous Calls
For outbound calls with privacy (anonymous) enabled, Avaya IP Office will replace the calling party
number in the From and Contact headers of the SIP INVITE message with “restricted” and
“anonymous” respectively. Avaya IP Office can be configured to use the P-Preferred-Identity (PPI)
or P-Asserted-Identity (PAI) header to pass the actual calling party information for authentication
and billing. For the compliance test, PAI was used for the purposes of privacy.
To configure Avaya IP Office to use PAI for privacy calls, navigate to User  noUser in the
Navigation / Group Panes. Select the Source Numbers tab in the Details Pane. Click the Add
button.
At the bottom of the Details Pane, the Source Number field will appear. Enter
SIP_USE_PAI_FOR_PRIVACY. Click OK.
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The SIP_USE_PAI_FOR_PRIVACY parameter will appear in the list of Source Numbers as
shown below.
5.9. SIP Options
Avaya IP Office sends SIP OPTIONS messages periodically to determine if the SIP connection is
active. The rate at which the messages are sent is determined by the combination of the Binding
Refresh Time (in seconds) set on the Network Topology tab in Section 5.1 and the
SIP_OPTIONS_PERIOD parameter (in minutes) that can be set on the Source Number tab of the
noUser user. The OPTIONS period is determined in the following manner:



If no SIP_OPTIONS_PERIOD parameter is defined and the Binding Refresh Time is 0,
then the default value of 44 seconds is used.
To establish a period less than 42 seconds, do not define a SIP_OPTIONS_PERIOD
parameter and set the Binding Refresh Time to a value less than 42 secs. The OPTIONS
message period will be equal to the Binding Refresh Time.
To establish a period greater than 42 seconds, a SIP_OPTIONS_PERIOD parameter must
be defined. The Binding Refresh Time must be set to a value greater than 42 secs. The
OPTIONS message period will be the smaller of the Binding Refresh Time and the
SIP_OPTIONS_PERIOD.
To configure the SIP_OPTIONS_PERIOD parameter, navigate to User  noUser in the
Navigation / Group Panes. Select the Source Numbers tab in the Details Pane. Click the Add
button.
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At the bottom of the Details Pane, the Source Number field will appear. Enter
SIP_OPTIONS_PERIOD=X, where X is the desired value in minutes. Click OK.
The SIP_OPTIONS_PERIOD parameter will appear in the list of Source Numbers as shown
below. For the compliance test, an OPTIONS period of 1 minute was desired. The Binding
Refresh Time was set to 60 seconds (1 minute) in Section 5.1. The SIP_OPTIONS_PERIOD
was set to 2 minutes. Avaya IP Office chose the OPTIONS period as the smaller of these two
values (1 minute). Click the OK button (not shown).
5.10. Save Configuration
Navigate to File  Save Configuration in the menu bar at the top of the screen to save the
configuration performed in the preceding sections (not shown).
6. Cbeyond SIP Trunking Configuration
Cbeyond is responsible for the configuration of the BeyondVoice sipConnect SIP Trunking service.
The customer will need to provide the IP address used to reach the Avaya IP Office at the local site.
Cbeyond will provide the customer the necessary information to configure the Avaya IP Office SIP
trunk connection to Cbeyond. The provided information from Cbeyond includes:
 IP address of the Cbeyond SIP proxy.
 Domain for Cbeyond’s BeyondVoice sipConnect service
 Supported codecs
 DID numbers
 IP addresses and port numbers used for signaling and media through any security devices.
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7. Verification Steps
The following steps may be used to verify the configuration:

Use the Avaya IP Office System Status application to verify the state of the SIP connection.
Launch the application from Start  Programs  IP Office  System Status on the PC
where Avaya IP Office Manager was installed. Select the SIP line of interest from the left
pane. On the Status tab in the right pane, verify that the Current State is Idle for each
channel (assuming no active calls at present time).

Select the Alarms tab and verify that no alarms are active on the SIP line.
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


Verify that a phone connected to the PSTN can successfully place a call to the Avaya IP
Office system with two-way audio.
Verify that a phone connected to Avaya IP Office can successfully place a call to the PSTN
with two-way audio.
Using a network sniffing tool (e.g., Wireshark), monitor the SIP signaling messages between
Cbeyond and Avaya IP Office on the REGISTER message from the local site to CBeyond for
registering the SIP trunk, and verify the SIP signaling message exchanges for Digest
Authentication:
1. Cbeyond SIP Trunking service returned a “401 Unauthorized” status message to the
initial REGISTER from Avaya IP Office. The 401 message contained a WWWAuthenticate Header posing challenge for Digest Authentication.
Example of WWW-Authenticate Header:
WWW-Authenticate: DIGEST
qop="auth",nonce="BroadWorksXgvjnm7ujThiysy2BW",algorithm=MD5,
realm="BroadWorks"
2. Avaya IP Office then presented the Digest Authentication response by sending a second
REGISTER that contained an Authorization Header supplying the information for
successful Digest Authentication. Note the username as configured in Section 5.4 (SIP
Credentials tab).
Example of Authorization Header:
Authorization: Digest
username="4167751396",realm="BroadWorks",nonce="BroadWorksXgvjnm7ujThiy
sy2BW",response="c061129dd6a93c7a0376aa9dfff93e76",uri="sip:sipconnectfca.atl0.cbeyond.net",algorithm=MD5,qop=auth,nc=00000001,cnonce="250811
0c8dee4adee79b"
3. CBeyond SIP Trunking service returned “200 OK” to signal successful registration of the
SIP trunk.
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
Using a network sniffing tool (e.g., Wireshark), monitor the SIP signaling messages between
Cbeyond and Avaya IP Office on an outbound call from the local site to the PSTN, and
verify the SIP signaling message exchanges for Digest Authentication:
1. Cbeyond SIP Trunking service returned a “401 Unauthorized” status message to the
initial INVITE from the Avaya IP Office. The 401 message contained a WWWAuthenticate Header posing challenge for Digest Authentication.
Example of WWW-Authenticate Header:
WWW-Authenticate: DIGEST
qop="auth",nonce="BroadWorksXgvjw1l9jThzk8gfBW",algorithm=MD5,
realm="BroadWorks"
2. Avaya IP Office ACKed the above 401 message, then presented the Digest
Authentication response by sending a second INVITE that contained an Authorization
Header supplying the information for successful Digest Authentication. Note the
username as configured in Section 5.4 (SIP Credentials tab).
Example of Authorization Header:
Authorization: Digest
username="4167751396",realm="BroadWorks",nonce="BroadWorksXgvjw1l9jThzk
8gfBW",response="5515bb16966de80df97b746fd70eeb73",uri="sip:19085551212
@sipconnect-fca.atl0.cbeyond.net:5060",algorithm=MD5,qop=auth,
nc=00000001,cnonce="e3f30cf6c14dc0a20b51"
3. CBeyond SIP Trunking service returned “100 Trying” and subsequent 18X call ringing or
session progress messages, signaling normal call progression.
8. Conclusion
Cbeyond SIP Trunking service passed compliance testing. These Application Notes describe the
procedures required to configure the SIP connection between Avaya IP Office and the Cbeyond SIP
Trunking service as shown in Figure 1.
9. Additional References
[1]
[2]
[3]
[4]
[5]
[6]
IP Office 7.0 IP Office Standard Version Installation, Document number15-601042, May 2011.
IP Office Release 7.0 Manager 9.0, Document number15-601011, May 2011.
IP Office Release 7.0 Voicemail Pro Administration, Document Number 15-601063, May 2011.
IP Office Release 6.0 System Status Application, Document number15-601758, February 2010.
IP Office System Monitor, Document Number 15-601019, November 28, 2008
IP Office Softphone Installation, Issue 2d, September 2011.
Product documentation for Avaya products may be found at http://support.avaya.com. Additional IP
Office documentation can be found at:
http://marketingtools.avaya.com/knowledgebase/
Product documentation for Cbeyond SIP Trunking is available from Cbeyond.
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Appendix: SIP Line Template
Avaya IP Office Release 7.0 supports SIP Line Template (in xml format) that can be created from an
existing configuration and imported into a new installation to simplify configuration procedures, as
well as to reduce potential configuration errors.
Note that not all of the configuration information, particularly items relevant to specific installation
environment, is included in the SIP Line Template. Therefore it is critical that the SIP Line
configuration be verified/updated after a template has been imported and additional configuration be
supplemented using Section 5.4 in these Application Notes as a reference.
The SIP Line Template created from the configuration as documented in these Application Notes is
as follows (with ITSP Proxy IP address masked and SIP Credentials for Digest Authentication
removed):
<?xml version="1.0" encoding="utf-8"?>
<Template xmlns="urn:SIPTrunk-schema">
<TemplateType>SIPTrunk</TemplateType>
<Version>20111207</Version>
<SystemLocale>enu</SystemLocale>
<DescriptiveName>US_Cbeyond_SIPTrunk</DescriptiveName>
<ITSPDomainName>sipconnect-fca.atl0.cbeyond.net</ITSPDomainName>
<SendCallerID>CallerIDNone</SendCallerID>
<ReferSupport>true</ReferSupport>
<ReferSupportIncoming>1</ReferSupportIncoming>
<ReferSupportOutgoing>1</ReferSupportOutgoing>
<RegistrationRequired>false</RegistrationRequired>
<UseTelURI>false</UseTelURI>
<CheckOOS>true</CheckOOS>
<CallRoutingMethod>1</CallRoutingMethod>
<OriginatorNumber />
<AssociationMethod>SourceIP</AssociationMethod>
<ITSPProxy>207.236.222.111</ITSPProxy>
<LayerFourProtocol>SipUDP</LayerFourProtocol>
<SendPort>5060</SendPort>
<ListenPort>5060</ListenPort>
<DNSServerOne>0.0.0.0</DNSServerOne>
<DNSServerTwo>0.0.0.0</DNSServerTwo>
<CallsRouteViaRegistrar>true</CallsRouteViaRegistrar>
<SeparateRegistrar />
<CompressionMode>AUTOSELECT</CompressionMode>
<UseAdvVoiceCodecPrefs>true</UseAdvVoiceCodecPrefs>
<AdvCodecPref>G.711 ULAW 64K</AdvCodecPref>
<CallInitiationTimeout>4</CallInitiationTimeout>
<DTMFSupport>DTMF_SUPPORT_RFC2833</DTMFSupport>
<VoipSilenceSupression>false</VoipSilenceSupression>
<ReinviteSupported>true</ReinviteSupported>
<FaxTransportSupport>FOIP_NONE</FaxTransportSupport>
<UseOffererPrefferedCodec>false</UseOffererPrefferedCodec>
<CodecLockdown>false</CodecLockdown>
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<T38FaxVersion>3</T38FaxVersion>
<Transport>UDPTL</Transport>
<LowSpeed>0</LowSpeed>
<HighSpeed>0</HighSpeed>
<TCFMethod>Trans_TCF</TCFMethod>
<MaxBitRate>FaxRate_14400</MaxBitRate>
<EflagStartTimer>2600</EflagStartTimer>
<EflagStopTimer>2300</EflagStopTimer>
<UseDefaultValues>true</UseDefaultValues>
<ScanLineFixup>true</ScanLineFixup>
<TFOPEnhancement>true</TFOPEnhancement>
<DisableT30ECM>false</DisableT30ECM>
<DisableEflagsForFirstDIS>false</DisableEflagsForFirstDIS>
<DisableT30MRCompression>false</DisableT30MRCompression>
<NSFOverride>false</NSFOverride>
<SIPCredentials>
<Expiry>60</Expiry>
<RegistrationRequired>true</RegistrationRequired>
</SIPCredentials>
</Template>
To import the above template into a new installation:
1. Copy and paste the above template into a text document named
US_Cbeyond_SIPTrunk.xml on the PC where IP Office Manager was installed. Move the
.xml file to the IP Office Manager template directory (C:\Program Files\Avaya\IP
Office\Manager\Templates).
2. Import the template into an IP Office installation by creating a new SIP Line as shown in the
screenshot below (right clicking Line in the left navigation pane):
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3. Verify that “United States” is automatically populated for Country and “Cbeyond” was
automatically populated for Service Provider in the resulting Template Type Selection
screen as shown below. Click Create new SIP Trunk to finish the importing process.
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©2012 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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