Configuring SIP Trunking between AT&

Avaya Solution & Interoperability Test Lab
Configuring SIP Trunking between AT&T IP Flexible
Reach and IP Toll Free Services with the Avaya Meeting
Exchange S6200 Conferencing Server via Avaya SIP
Enablement Services - Issue 1.0
Abstract
These Application Notes present the procedures for configuring SIP Trunking connectivity
between the AT&T IP Flexible Reach and IP Toll Free services with the Avaya Meeting
Exchange S6200 Conferencing Server via Avaya SIP Enablement Services.
AT&T IP Flexible Reach and IP Toll Free are managed Voice over IP communication
solutions using SIP trunks to provide inbound and outbound local, long distance, international
and toll free services for U.S. sites.
AT&T is a member of the Avaya DevConnect Service Provider program. Information in these
Application Notes has been obtained through compliance testing and additional technical
discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and
Interoperability Test Lab in conjunction with remote access to the AT&T Virtual
Interoperability Test lab.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
1 of 43
ATT-S6200
1. Introduction
These Application Notes present the procedures for configuring SIP trunking connectivity
between the AT&T IP Flexible Reach and IP Toll Free services and the Avaya Meeting
Exchange S6200 Conferencing Server via Avaya SIP Enablement Services.
AT&T IP Flexible Reach and IP Toll Free services (AT&T Services) are managed Voice over IP
communication solutions using SIP trunks to provide inbound and outbound local, long distance,
international and toll free services for U.S. sites.
Avaya Meeting Exchange is an advanced conferencing solution offering reservation-less and
scheduled meet-me voice conferencing capabilities using SIP trunking.
The following conferencing features have been verified:
• Dial-In Conferencing:
o Dialed Number Identification Service (DNIS) direct call function, where
conference participants enter a conference as moderator without entering a
participant access code (passcode).
o Scan call function, where conference participants enter a conference with a valid
passcode.
• Dial-Out Conferencing from Avaya Meeting Exchange:
o Blast dial
ƒ Auto, where a conference participant enters a conference via a DNIS
direct call function and automatically invokes a Blast dial to a preprovisioned dial list of one or more participants.
ƒ Manual, where a conference participant is already in a conference as a
moderator and invokes a Blast dial to a pre-provisioned dial list of one or
more participants.
o Originator Dial-Out, where a conference participant is already in a conference as
a moderator and invokes a Dial-Out to a single participant.
o Operator Fast Dial, where an operator can Dial-Out to a pre-provisioned dial list
of one or more participants.
• Operator Dial-Out to set up an Audio Path.
• Operator Dial-In to set up an Audio Path.
• Dial-Out for audio recording.
• Line Transfer initiated from Avaya Bridge Talk.
• Conference Transfer initiated from Avaya Bridge Talk.
The following codecs were verified:
• G.711mu
(Note: The Avaya S6200 Meeting Exchange only supports the G.711mu and G.711a
codecs; G.729 is not available.)
Fax is not supported by the Avaya S6200 Meeting Exchange and is not applicable to these tests.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
2 of 43
ATT-S6200
These Application Notes provide the administrative steps for configuring the application as
shown in Figure 1.
1
2
4
3
5
7
#
6
8
0
9
*
1
2
4
3
5
7
#
6
8
0
9
*
Figure 1: Network Configuration
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
3 of 43
ATT-S6200
1.1. AT&T Services Configuration Information
These Application Notes provide an illustrative example of how the Avaya S6200 Meeting
Exchange is configured with the AT&T IP Flexible Reach and IP Toll Free services.
The specific values provided below are illustrative only and must not be used for customer
configurations. Each customer must obtain the specific values for their configuration from
AT&T during service provisioning of their AT&T IP Flexible Reach or IP Toll Free services.
AT&T Provisioning Information
AT&T Border Element Address(es)
G.711MU Codec Support
RFC 2833 (DTMF Event) Supported
Assigned Direct Inward Dial (DID) Numbers
DID Digits Passed in SIP Request URI
DID Digits Passed in SIP To Header
Incoming Toll Free Number
Incoming Toll Free Digits Passed in SIP Request URI
Incoming Toll Free Digits Passed in SIP To Header
Illustrative Values in these
Application Notes
210.245.228.200
210.245.118.100
Yes
Yes
1-536-555-0442
1-536-555-0443
5365550442
5365550443
Same as SIP Request URI
800-457-5711
0000000010
0004152100010
2. Equipment and Software Validated
The following equipment and software versions were used for the configuration:
Equipment
Avaya Meeting Exchange S6200 Conferencing
Server
Software
Release 5.0
Avaya Meeting Exchange Bridge Talk
Release 5.0
Avaya SIP Enablement Services
SES-4.0.0.0-033.6
AT&T IP Flexible Reach and IP Toll Free
VNI-9
Services
Table 1 - Hardware and Software Versions
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
4 of 43
ATT-S6200
3. Avaya Meeting Exchange Configuration
This section describes the steps for configuring Avaya Meeting Exchange to interoperate with
Avaya SIP Enablement Services via secure SIP connectivity utilizing Transport Layer Security
(TLS).
Step
1
2
Description
Log in to the Avaya Meeting Exchange Server console with the appropriate credentials.
Configure settings that enable secure SIP connectivity between Avaya Meeting Exchange and
other SIP User Agents by editing the system.cfg file as follows:
• cd to /usr/ipcb/config.
• Edit the system.cfg file with a text editor, e.g., vi.
• Add a line to identify the IP address of Avaya Meeting Exchange (as defined in the
/etc/hosts file), e.g.,
o IPAddress=144.153.9.227
• Add a line to populate the From header field in SIP INVITE messages from Avaya
Meeting Exchange, e.g.,
MyListener=sip:6000@144.153.9.227
The string “6000” is arbitrarily chosen.
• Add a line to provide User Agents a Contact address to use for acknowledging SIP
messages from Avaya Meeting Exchange, e.g.,
o respContact=<sip:6000@144.153.9.227:5061;transport=tls>
Note: The string “6000” is arbitrarily chosen.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
5 of 43
ATT-S6200
Step
3
Description
To associate incoming calls to Avaya Meeting Exchange with different call handling flows,
edit the UriToTelnum.tab file to extract the Direct Inward Dial / Dialed Number
Identification Service (DID/DNIS) and Automatic Number Identification (ANI) values as
follows:
• cd to /usr/ipcb/config.
• Edit the UriToTelnum.tab file with a text editor, e.g., vi.
o Add a line to match the regular expression pattern of the To and From headers
in SIP INVITE messages from the AT&T services. In these Application Notes
this line is:
"<sip:*@*"
$1
If a match occurs, the $1 variable will contain the DID/DNIS address digits
extracted from the To header and the ANI extracted from the From header.
For example, “5365550442” is the DID/DNIS value derived from the following
To header.
To: <sip:5365550442@15.163.182.124;user=phone>
and “+17358551637” is the ANI value derived from the following From header.
From: "John" <sip:+17358551637@210.245.228.200:5060;user=phone>;
•
Enable an undefined caller to receive a prompt for operator assistance by administering
for the condition of an unmatched SIP INVITE message by adding a wildcard entry as
the last line in this file.
o This line is:
*
$0
Note: Entries in this file are read sequentially and the first match used;
therefore, the undefined caller line (e.g., * $0 ) must be the last line in the
file. Otherwise, all calls to Avaya Meeting Exchange would match the wildcard
and thus receive a prompt for operator assistance.
Specific guidelines for the configuration of this table are discussed in Chapter 3 of Reference
[1].
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
6 of 43
ATT-S6200
Step
4
Description
To enable Dial-Out from Avaya Meeting Exchange using SIP trunking to the AT&T services
via the Avaya SES, edit the telnumToUri.tab file as follows:
• cd to /usr/ipcb/config.
• Edit the telnumToUri.tab file with a text editor, e.g., vi.
• Add a line to the file to route all outbound calls from Avaya Meeting Exchange to
Avaya SIP Enablement Services, e.g.,
* sip:$0@15.163.182.124:5061;transport=tls default_gateway
In this example:
ƒ the pattern “*” is a wild card that matches any dialed digits,
ƒ the string “sip:$0@15.163.182.124:5061;transport=tls” is the URI that
will be sent to the Avaya SES (at 15.163.182.124) using the tls transport
protocol on port 5061. Avaya Meeting Exchange will replace “$0” with
the actual dialed digits.
ƒ the string “default_gateway” is a comment describing the purpose of the
line.
3.1. Call Routing Configuration
The following steps configure the Avaya Meeting Exchange to handle the expected length of the
DID/DNIS digits received from AT&T on incoming calls. This prepares the CBUTIL utility
(covered in the next section) to perform the appropriate searches in the CBUTIL call branding
tables.
Call Routing Configuration is performed using the System Management Interface.
Step
5
Description
To configure Call Routing Configuration using the System Management Interface:
• Log into the Avaya Meeting Exchange Server console with the appropriate credentials.
• At the command prompt, enter “dcbadmin 116” to invoke the System Management
Interface.
[s6200 ]# dcbadmin 116
The System Administrator Main Menu page should appear.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
7 of 43
ATT-S6200
Step
6
7
Description
Select the Configurations menu item and press Enter.
Select Call Routing Configuration and press Enter.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
8 of 43
ATT-S6200
Step
8
Description
Select Digit Parameters and press Enter.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
9 of 43
ATT-S6200
Step
9
Description
Set the Number of Digits to the maximum length incoming digit string (DNIS) expected.
In these Application Notes the AT&T IP Toll Free service sends 13 digits and the AT&T IP
Flexible Reach service sends 10 digits. Thus, set the Number of Digits value to “13”.
Since the Number of Digits varies for the two AT&T services, set the Short Collection
Search value to “ON”. This instructs the Avaya Meeting Exchange to attempt a partial match
(in right to left order) in the call branding table when fewer then 13 DNIS digits are received.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
10 of 43
ATT-S6200
Step Description
10 Press “ESC” and “Y” to save the Digit Parameters changes.
11 Reboot Avaya Meeting Exchange for changes to take effect.
Note: Rebooting Avaya Meeting Exchange is service impacting.
[S6200]> init 6
3.2. CBUTIL Utility
The CBUTIL utility enables specific annunciator messages, line name, company name and
routing function to be assigned to each DID/DNIS patterns. These assignments are stored in the
call branding table. The DID/DNIS values are obtained from the To Header of the SIP INVITE
messages according to the rules specified in Step 3. Note that the values in the To header may
not match the digits found in the SIP Request URI.
The routing functions used in these Application Notes are:
• ENTER – places the incoming call matching the corresponding DID/DNIS pattern into an
Avaya Meeting Exchange ENTER queue for handling by an operator. The operator will
screen the call and place the caller into the proper conference using the BridgeTalk
application.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
11 of 43
ATT-S6200
•
•
DIRECT – places the incoming calls matching the DID/DNIS pattern directly into an
assigned conference without operator screening or caller entered access codes.
SCAN – prompts caller to enter a conference access code before for placing them into the
conference matching the DID/DNIS patterns. Failed attempts are routed to the ENTER
queue for operator handling.
In these Application Notes the AT&T Services DID/DNIS numbers are assigned the Avaya
Meeting Exchange routing function as shown in Table 2.
Dialed PSTN
Number
1-800-457-5711
1-536-555-0442
1-536-555-0443
Other
Digits Received
in the SIP
Request URI
0000000010
5365550442
5365550443
DID/DNIS Digits
Avaya Meeting
Received in SIP To
Exchange Assigned
Header
Routing Function
0004152100010
SCAN
5365550442
SCAN
5365550443
DIRECT
Other Unrecognized
ENTER
Table 2 - Call Branding Routing Function Assignments
Step Description
12 To provide the call branding treatment defined in Table 2 using the DID/DNIS values obtained
by the rule defined in Step 3, run the cbutil utility as follows:
• Log in to the Avaya Meeting Exchange Server console with the appropriate credentials.
• At the command prompt, run the cbutil utility to administer DNIS entries provisioned
on Avaya Meeting Exchange.
Note that entering cbutil without an additional command argument displays the cbutil
help. Entering cbutil list will list all existing entries in the call branding table.
[craft@s6200 ~]$ cbutil
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
Usage: <command> [command-specific args...]
where <command> may be one of:
add
Add an entry to the Call Branding table
remove
Remove an entry from the Call Branding table
update
Update an entry in the Call Branding table
lookup
Display an entry in the Call Branding table
count
Display the number of entries in the Call Branding table
list
List entries in the Call Branding table
dnissize
Set system configured max dnis length (1-16)
Note: This command should only be used when the bridge is not running.
Use "<command> -help" to get help on a specific command
[craft@s6200 ~]$
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
12 of 43
ATT-S6200
Step Description
13 Enable dial-in access (via passcode) to conferences using the AT&T Toll Free DID/DNIS
value “0004152100010” with the following command. These conferences use the SCAN call
routing function.
cbutil add <dnis> <rg> <msg> <ps> <ucps> <func> [-l <ln> -c <cn>], where the
variables for add command are defined as follows:
o <dnis>
DNIS
o <rg>
Reservation Group
o <msg>
Annunciator message number
o <ps>
Prompt Set number (0-20)
o <ucps>
Use Conference Prompt Set (y/n)
o <func>
One of: DIRECT/SCAN/ENTER/HANGUP/AUTOVL/FLEX
o -l <"ln"> Optional line name to associate with caller
o -c <"cn"> Optional company name to associate with caller
[craft@s6200 ~]$ cbutil add 0004152100010 0 1 1 n scan
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
[craft@s6200 ~]$
14 In a similar manner, enable dial-in access (via passcode) to conferences using the AT&T IP
Flexible Reach “5365550442” DID/DNIS value. These conferences use the SCAN call
routing function.
Note that the <dnis> value must be padded using the “?” wild card character to the full 13 digit
length previously set in Section 3.1.
[craft@s6200 ~]$ cbutil add ???5365550442 0 1 1 n scan
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
[craft@s6200 ~]$
15 Enable dial-in access (without entering a passcode) using the AT&T IP Flexible Reach
“5365550443” DID/DNIS value. These conferences use the DIRECT call routing function.
Note that the <dnis> value must be padded using the “?” wild card character to the full 13 digit
length previously set in Section 3.1.
[craft@s6200 ~]$ cbutil add ???5365550443 0 301 1 n direct
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
[craft@s6200 ~]$
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
13 of 43
ATT-S6200
Step Description
16 At the command prompt, enter cbutil list to verify the DNIS entries were administered
correctly.
Note: The last entry in the call brand table is the wild card entry “???”. This entry captures
any wrong number (e.g., unmatched DID/DNIS values) and places the call into the Enter
queue for operator assistance.
[craft@s6200 ~]$ cbutil list
cbutil
Copyright 2004 Avaya, Inc. All rights reserved.
DNIS
---------------0004152100010
???5365550442
???5365550443
?????????????
[craft@s6200 ~]$
JSR; Reviewed:
SPOC 4/10/2008
Grp
--0
0
0
0
Msg
--1
1
301
208
PS
--1
1
1
1
CP
-N
N
N
N
Function Line Name
Company Name
-------- -------------------- ----------------SCAN
SCAN
DIRECT
ENTER
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
14 of 43
ATT-S6200
3.3. Bridge Talk
The following steps provide an example of how to provision a conference on Avaya Meeting
Exchange from the Avaya Bridge Talk application. This sample conference is utilized in
conjunction with the Direct and Scan call functions (provisioned in the previous section) to
enable both Dial-In and Dial-Out access to audio conferencing for endpoints associated with
Avaya Communication Manager.
Note: If any of the features shown in the following Avaya Bridge Talk screen captures are not
present, contact an authorized Avaya sales representative to make the appropriate changes.
Step Description
17 Open the Avaya Bridge Talk application and log in to Avaya Meeting Exchange with the
appropriate credentials.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
15 of 43
ATT-S6200
Step Description
18 Provision a dial list that is utilized for Dial-Out (e.g., Blast dial and Fast Dial) from Avaya
Meeting Exchange.
From the Avaya Bridge Talk menu bar, click Fast Dial Î New.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
16 of 43
ATT-S6200
Step Description
19 From the New Dial List window that is displayed:
• Enter a descriptive name for the Name field.
• Add entries to the dial list by clicking the Add button for each entry.
o Assign moderator privileges to a conference participant by checking the
Moderator box.
• See Reference 3 in Section 8 for provisioning of the remaining entries in this screen.
• When finished, click the Save button on the bottom of the screen.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
17 of 43
ATT-S6200
Step Description
20 Provision a conference with Auto Blast enabled.
From the Avaya Bridge Talk menu bar, click View Î Conference Scheduler.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
18 of 43
ATT-S6200
Step Description
21 From the Conference Scheduler window that is displayed, click File Î Schedule
Conference.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
19 of 43
ATT-S6200
Step Description
22 From the Schedule Conference window that is displayed, provision a conference as follows:
• Enter a unique Conferee Code to allow participants access to this conference.
• Enter a unique Moderator Code to allow participants access to this conference with
moderator privileges.
• Enter a descriptive name for the Conference Name field.
• Administer settings to enable an Auto Blast dial by setting Auto Blast to Auto and
selecting the dial list provisioned in Step 19.
o [Not Shown] Select a dial list by clicking the Dial List button, then selecting a
dial list from the Create, Select or Edit Dial List window that is displayed and
clicking the Select button.
• See Reference 3 in Section 8 for provisioning of the remaining entries in this screen.
• When finished, click the OK button on the bottom of the screen.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
20 of 43
ATT-S6200
4. Avaya SIP Enablement Services Configuration
This section describes the steps for configuring Avaya SIP Enablement Services to serve as a SIP
proxy between the Avaya Meeting Exchange and the AT&T SIP trunking services.
4.1. Initial Avaya SES Setup
These Application Notes assume the Avaya SES is configured immediately following the initial
installation (using the “initial_setup” script). The SIP Server Management page “setup” steps
may vary slightly if the Avaya SES has been previously administered. If so, the left hand
navigation menu of the SIP Server Management pages may be used to locate the referenced
pages as appropriate.
Step
1
2
Description
Administer settings for Avaya SIP Enablement Services as follows:
• Open a web browser and enter the following URL:
https://<IP address of Avaya SIP Enablement Services>/admin
• Log in to Avaya SIP Enablement Services with the appropriate credentials.
Click the Launch Administration Web Interface link to enter the SIP Server Management
pages.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
21 of 43
ATT-S6200
Step
3
Description
From the main SIP Server Management page:
• Click on the Setup link to start the initial configuration.
Note: These Application Notes assume the Avaya SES is being configured immediately
following installation. In this case, the Setup link will be present. If not, in the
following steps access the corresponding page directly using the left hand navigation
menus.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
22 of 43
ATT-S6200
Step
4
Description
Follow the setup links to reach the View System Properties page.
• Enter the customer’s SIP Domain and License Host values as shown (if not previously
configured).
• Click Update when done.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
23 of 43
ATT-S6200
Step
5
Description
Follow the links to reach the Add Host page.
• Enter the Avaya SES IP address in the Host IP Address field.
• Enter the Avaya SES database password (assigned when the “initial_setup” installation
script was run) in the DB Password field.
• Enter a password that uniquely identifies the Avaya SES for intra (and inter) proxy
communications in the Profile Service Password field.
• Select “TLS” from the available Link Protocols, which is consistent with the
“system.cfg” file configured for Avaya Meeting Exchange in Section 3 Step 2.
• Leave all remaining fields at their default settings as shown.
• Click the Add button when done.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
24 of 43
ATT-S6200
Step
6
Description
Follow the links to reach the Add Media Server Interface page.
To configure the Avaya Meeting Exchange as a media server:
• Enter a descriptive name for the Avaya Meeting Exchange in the Media Server
Interface Name field.
• Select “TLS” as the SIP Trunk Link Type consistent with the Avaya Meeting
Exchange “system.cfg” file configuration done in Section 3 Step 2.
• Enter the IP address of the Avaya Meeting Exchange in the SIP Trunk IP Address
field.
• Leave the Media Server Admin fields blank.
• Select “Telnet” in the SMS Connection Type field.
• Click the Add button when done.
4.2. Dial-In to the Avaya Meeting Exchange
The following steps configure the Avaya SES to route incoming calls to the Avaya Meeting
Exchange. In these Application Notes, two Direct Inward Dialed numbers (1-536-555-0442, 1JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
25 of 43
ATT-S6200
536-555-0443) and one Toll Free number (1-800-457-5711 with the incoming DNIS digits of
“0000000010”) have been provided by the AT&T Services. (See Table 2)
Step
7
8
Description
On the List Media Servers page:
• Click the Map link to add the address maps that will route the incoming calls from the
AT&T Services to the Avaya Meeting Exchange.
On the List Media Server Address Map page:
• Click the Add Map In New Group link.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
26 of 43
ATT-S6200
Step
9
Description
On the Add Media Server Address Map page:
• Enter a descriptive name for the incoming number in the Name field. Here, “ATTDID” indicates that this map is used for the incoming Direct Inward Dialed numbers
assigned by the AT&T IP Flexible Reach Service.
• Enter in the Pattern field the regular expression pattern matching the incoming call
digits received from the AT&T Services in the SIP INVITE message.
In these Application Notes the pattern “^sip:536555044[23]” is used. This indicates
that a Request URI beginning with “sip:” followed by the 9 digits “536555044” and
either “2” or “3” will be matched and routed to the Avaya Meeting Exchange SIP
Trunk IP address defined in Section 4.1 Step 6.
The corresponding portion (shown in bold) of a Request URI that this pattern matches
is: “INVITE sip:5365550442@15.163.182.124:5060 SIP/2.0”.
•
•
Check the Replace URI field.
Click the Add button when done.
Note: The regular expression patterns may be designed to match more than one incoming DID
number by using additional types of matching patterns. The further use of regular expression
patterns is described in Appendix B.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
27 of 43
ATT-S6200
Step Description
10 On the List Media Server Address Map page:
• Click the Add Another Map link to create a second incoming call address map for the
AT&T Services toll free number.
11 Enter the information for the AT&T Services toll free number as was done in Step 9.
Note in this case the AT&T DNIS digits of “0000000010” are what is received in the incoming
Request URI.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
28 of 43
ATT-S6200
Step Description
12 The resulting List Media Server Address Map page displays the completed incoming call
address routing configuration. Note that the Contact information displays the Request URI
that will be used to communicate with the Avaya Meeting Exchange.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
29 of 43
ATT-S6200
4.3. Dial-Out from the Avaya Meeting Exchange
The following steps configure the Avaya SES to route outbound calls from the Avaya Meeting
Exchange to the AT&T IP Flexible Reach Service for completion to a Public Switched
Telephone Number (PSTN) telephone number.
Step Description
13 From any SIP Server Management page:
• Click the List link under the Hosts section of the left navigation bar.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
30 of 43
ATT-S6200
Step Description
14 From the List Hosts page:
• Click the Map link to display the current Host Address Maps.
15 From the List Host Address Map page:
• Click the Add Another Map link.
Note: Use the Add Map in New Group link if there is a Contact previously defined
(that does not correspond to the AT&T Border Element Address).
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
31 of 43
ATT-S6200
Step Description
16 From the Add Host Address Map page:
• Enter a descriptive name for the outbound address map in the Name field. Here,
“ATT-DialOut-LD” indicates this will be used for 1+10 digits PSTN calls routed via
the AT&T IP Flexible Reach Service.
• Enter in the Pattern field the regular expression pattern matching the outgoing call
digits being sent to the AT&T IP Flexible Reach Service in the SIP INVITE message.
In these Application Notes the pattern “^sip:1[0-9]{10}” is used. This indicates that a
Request URI beginning with “sip:” plus “1” plus any 10 digits will be matched and
routed to the associated Contact to be defined in the next step.
The corresponding portion (shown in bold) of the Request URI that this pattern
matches is “INVITE sip:17358551637@210.245.228.200:5060;transport=udp
SIP/2.0”.
•
•
JSR; Reviewed:
SPOC 4/10/2008
Check the Replace URI field.
Click the Add button when done.
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
32 of 43
ATT-S6200
Step Description
17 From the List Host Address Map page:
• Click the Add Another Contact link corresponding to the ATT-DialOut-LD address
map created in the previous step.
18 From the Add Host Contact page:
• Enter the required SIP Request URI that will be sent to the AT&T IP Flexible Reach
Service. In the case, the specific entry (without the double quotes) is:
“sip:$(user)@210.245.228.200:5060;transport=udp”
This indicates that the SIP INVITE will be sent to the AT&T Border Element Address
(202.242.225.200) using port 5060 and the udp transport method. The “$(user)” is a
variable used to substitute the specific dialed address (e.g., 17358551637) used for each
call.
•
JSR; Reviewed:
SPOC 4/10/2008
This information will vary for individual customers and must be obtained from AT&T
as part of the AT&T Services provisioning process.
Click the Add button when done.
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
33 of 43
ATT-S6200
Step Description
19 The List Host Address Map page is displayed.
• Verify the Contact information is properly entered and assocated with the correct
address pattern Name.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
34 of 43
ATT-S6200
4.4. AT&T Services as a Trusted Host
This section designates that the AT&T Border Element Addresses are trusted hosts to the Avaya
SES. This prevents the Avaya SES from challenging the AT&T Services for SIP authentication
when SIP messages are received from the AT&T Services.
20 Configure the Avaya SES trusted hosts by clicking the Add link under Trusted Hosts.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
35 of 43
ATT-S6200
21 The Add Trusted Host page is displayed.
To add the AT&T Services as a trusted host:
• Enter the AT&T Border Element Address in the IP Address field.
• Set the Host field to the Avaya SES IP address.
• Enter a descriptive phrase into the Comment field.
• Click the Add button when finished.
22 Repeat for any other AT&T Border Element Addresses provided.
23 Confirm the entries on the List Trusted Hosts page.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
36 of 43
ATT-S6200
4.5. Commit Avaya SES Administrative Changes
The various Avaya SES administrative changes performed above will not take effect until the
update action is performed.
Step Description
24 To perform the Avaya SES update action:
• Click on either Update link found any SIP Server Management page.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
37 of 43
ATT-S6200
5. Verification Steps
The following steps can be used to verify the configuration described in these Application Notes.
5.1. Verification Tests
This section provides steps that may be performed to verify the operation of the SIP trunking
configuration described in the Application Notes.
•
Incoming Calls – Verify that calls placed from a PSTN telephone using the AT&T
provided DID or toll free telephone number assigned are properly routed via the SIP
trunk to the Meeting Exchange. The expected Avaya Meeting Exchange announcement
should be heard. Verify that the conference PIN is accepted and/or calls are routed to the
Avaya Bridge Talk operator queue. Verify the talk-path exists in both directions, among
all various conference participants and that calls remain stable for several minutes and
disconnect properly.
•
Outbound Calls – Verify that an Avaya Bridge Talk operator or conference moderator
can place outbound calls to a PSTN destination via the AT&T Services. Verify that the
talk-path exists in both directions, among all various conference participants and that
calls remain stable and disconnect properly.
•
Using Avaya Bridge Talk verify participants in conferences, operator ability to monitor
and enter conferences, and the ability of the operator to add and disconnect conference
parties.
5.2. Troubleshooting Tools
The “Trace Logger” function within the Avava SES Administration Web Interface may be used
to capture SIP traces between Avaya SES and the AT&T Services. These traces can be
instrumental in understanding SIP protocol issues resulting from configuration problems.
If port monitoring is available, a SIP protocol analyzer such as WireShark (a.k.a., Ethereal) to
monitor the SIP messaging between the SES and the AT&T Services. Note that SIP messaging
between Avaya Meeting Exchange and Avaya SES uses TLS encryption and cannot be viewed
using WireShark.
6. Support
AT&T customers may obtain support for the AT&T IP Flexible Reach Service by calling 1-877288-8362. Support for the AT&T IP Toll Free Service should be directed to 1-800-325-5555.
Avaya customers may obtain documentation and support for Avaya products by visiting
http://support.avaya.com. The “Connect with Avaya” section provides the worldwide support
directory. In the United States, 1-866-GO-AVAYA (1-866-462-8292) provides access to overall
sales and service support menus. Customers may also use specific numbers (provided on
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
38 of 43
ATT-S6200
support.avaya.com) to directly access specific support and consultation services based upon their
Avaya support agreements.
7. Conclusion
These Application Notes provide administrators with the procedures to configure SIP trunking
connectivity between AT&T IP Flexible Reach and IP Toll Free services with the Avaya
Meeting Exchange S6200 Conferencing Server via Avaya SIP Enablement Services.
8. Additional References
The following Avaya references are available at http://support.avaya.com.
1. Meeting Exchange 5.0 Administration and Maintenance S6200/S6800 Media Server,
Issue 1, Doc ID 04-602167, August 2007.
2. Avaya Meeting Exchange Groupware Edition Version 4.1 User’s Guide for Bridge
Talk, Issue 2, Doc ID 04-600878, July 2006.
3. SIP Enablement Services Implementation Guide, Issue 4, Doc ID: 16-300140, May
2007.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
39 of 43
ATT-S6200
APPENDIX A: Sample SIP INVITE Messages
This section displays the format of typical SIP INVITE messages sent between AT&T and the
Avaya SES. These INVITE messages may be used for comparison and troubleshooting
purposes. Differences in these messages may indicate that different configuration options were
selected.
Sample SIP INVITE Message from the AT&T services to the Avaya SES:
INVITE sip:5365550442@144.153.9.227:5060;transport=tcp SIP/2.0
Accept: application/sdp,application/isup,application/dtmf,application/dtmf-relay,multipart/mixed
Accept-Language: en;q=0.0
Call-ID: SD5ah8801-0f2fa65682480bc12b825764c9126f03-fms3e43
CSeq: 1 INVITE
From: "OUT_OF_AREA" <sip:+17358551637@210.245.228.200:5060;user=phone>;tag=SD5ah8801-ds84f142e2
To: <sip:5365550442@15.163.182.124;user=phone>
Via: SIP/2.0/TCP 15.163.182.124:5060;branch=z9hG4bK263535C644441353534558.0,SIP/2.0/UDP
210.245.228.200:5060;psrrposn=1;received=210.245.228.200;branch=z9hG4bKbl156u1010g0lb0i17g1.1
Content-Length: 275
Content-Type: application/sdp
Contact: <sip:+17358551637@210.245.228.200:5060;transport=udp>
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Content-Disposition: session;handling=required
P-Asserted-Identity: <sip:7358551637@210.245.228.200:5060>
Record-Route: <sip:15.163.182.124:5060;transport=tcp;lr>
v=0
o=Sonus_UAC 5764 1479 IN IP4 210.245.228.200
s=SIP Media Capabilities
c=IN IP4 210.245.228.200
t=0 0
m=audio 17076 RTP/AVP 2 18 0 96
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
40 of 43
ATT-S6200
Sample SIP INVITE Message from Avaya SES to the AT&T services:
INVITE sip:17358551637@210.245.228.200:5060;transport=udp SIP/2.0
Call-ID: 7CC40F2A@144.153.9.2275060
CSeq: 833742147 INVITE
Expires: 180
From: <sip:6000@144.153.9.227>;tag=144.153.9.2275060+1+13d0000+b4c5d8d5
To: sip:17358551637@15.163.182.124:5060;transport=tcp
Via: SIP/2.0/UDP 15.163.182.124:5060;branch=z9hG4bKB273136334433634134218.0,SIP/2.0/TCP
144.153.9.227:5060;psrrposn=1;received=144.153.9.227;branch=z9hG4bK+6c07642345e3508f8cf868e049f50
bb9+144.153.9.2275060+1
Content-Length: 218
Content-Type: application/sdp
Contact: <sip:S6200@144.153.9.227:5060;transport=tcp>
Max-Forwards: 69
Supported: timer
Min-SE: 900
Session-Expires: 900
Record-Route: <sip:15.163.182.124:5060;lr>
v=0
o=- 534363200 534363200 IN IP4 144.153.9.227
s=c=IN IP4 144.153.9.227
t=0 0
m=audio 42044 RTP/AVP 0 8 101
a=rtpmap:8 pcma/8000/1
a=rtpmap:0 pcmu/8000/1
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
41 of 43
ATT-S6200
APPENDIX B: Specifying Pattern Strings in Address Maps
The syntax for the pattern matching used within the Avaya SES is a Linux regular expression
used to match against the URI string found in the SIP INVITE message.
Regular expressions are a way to describe text through pattern matching. The regular expression
is a string containing a combination of normal text characters, which match themselves, and
special metacharacters, which may represent items like quantity, location or types of
character(s).
In the pattern matching string used in the Avaya SES:
• Normal text characters and numbers match themselves.
• Common metacharacters used are:
o A period . matches any character once (and only once).
o An asterisk * matches zero or more of the preceding characters.
o Square brackets enclose a list of any character to be matched. Ranges are
designated by using a hyphen. Thus the expression [12345] or [1-5] both
describe a pattern that will match any single digit between 1 and 5.
o Curly brackets containing an integer ‘n’ indicate that the preceding character must
be matched exactly ‘n’ times. Thus 5{3} matches ‘555’ and [0-9]{10}
indicates any 10 digit number.
o The circumflex character ^ as the first character in the pattern indicates that the
string must begin with the character following the circumflex.
Putting these constructs together as used in this document, the pattern to match the SIP INVITE
string for any valid 1+ 10 digit number in the North American dial plan would be:
^sip:1[0-9]{10}
This reads as: “Strings that begin with exactly sip:1 and having any 10 digits following will
match.
A typical INVITE request below uses the shaded portion to illustrate the matching pattern.
INVITE sip:17325551638@20.1.1.54:5060;transport=udp SIP/2.0
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
42 of 43
ATT-S6200
©2008 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.
JSR; Reviewed:
SPOC 4/10/2008
Solution & Interoperability Test Lab Application Notes
©2008 Avaya Inc. All Rights Reserved.
43 of 43
ATT-S6200
Download PDF