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Abstract
These Application Notes describe the procedure for configuring the AudioCodes Mediant
1000 VoIP Media Gateway to interoperate with Avaya SIP Enablement Services and Avaya
Communication Manager.
The AudioCodes Mediant 1000 VoIP Media Gateway serves as a gateway between TDM and
IP networks. The Mediant 1000 supports multiple hardware interfaces and control protocols.
Capacity can be scaled upward by adding additional interface modules. The compliance test
configured the Mediant 1000 as a SIP to ISDN-PRI gateway connecting two enterprise sites.
The AudioCodes Mediant 1000 VoIP Media Gateway serves as a gateway between TDM and IP
networks. The Mediant 1000 supports multiple hardware interfaces and control protocols. Capacity
can be scaled upward by adding additional interface modules. The compliance test configured the
Mediant 1000 as a SIP to ISDN-PRI gateway connecting two enterprise sites.
1.1. Configuration
Figure 1 illustrates the configuration used in these Application Notes. In the sample configuration,
two sites are connected via an IP network. Located at Site 1 is an Avaya SES and an Avaya S8300
Media Server running Avaya Communication Manager in an Avaya G350 Media Gateway.
Endpoints include an Avaya 4600 Series IP Telephone with SIP firmware, an Avaya 4600 Series IP
Telephone with H.323 firmware, an Avaya 6408D Digital Telephone, and a fax machine. There is
no direct connection to the PSTN at this site.
Located at Site 2 is an AudioCodes Mediant 1000 VoIP Media Gateway connected to an Avaya
DEFINITY Server R via an ISDN-PRI trunk. The Avaya DEFINITY Server R in turn has an ISDN-
PRI connection to the PSTN. Site 2 provides PSTN access for both sites. All outbound calls from
any phone at Site 1 will be routed across the SIP connection starting at Avaya Communication
Manager at Site 1 through Avaya SES and terminating at the Mediant 1000 at Site 2. The Mediant
1000 routes the call from its SIP interface to its ISDN-PRI interface which is connected to the Avaya
DEFINITY Server R. The Avaya DEFINITY Server R then routes the call to the PSTN. Inbound
calls to Site 1 follow the same path in the reverse order, entering the network at the Avaya
DEFINITY Server R at Site 2.
The incoming PSTN numbers of the ISDN-PRI trunk are mapped to telephone extensions at Site 1.
Equipment Software/Firmware
Avaya S8300 Media Server with Avaya G350 Avaya Communication Manager 3.1.2
Media Gateway (R013x.01.2.635.0)
This is a post-GA build needed to
support T.38 fax with the Mediant
1000.
Avaya SIP Enablement Services (SES) 3.1 (build 18)
Avaya DEFINITY Server R R011r.03.1.535.0
Avaya 4620SW IP Telephones SIP version 2.2.2
Avaya 4620 IP Telephones H.323 version 2.3
Avaya 6408D Digital Telephone -
Analog Telephones -
Analog Fax Machines -
Windows PCs Windows XP Professional
AudioCodes Mediant 1000 VoIP Media 5.00A.011.008
Gateway
The license file installed on the system controls the maximum permitted. If a required
feature is not enabled or there is insufficient capacity, contact an authorized Avaya
sales representative to make the appropriate changes.
(NOTE: You must logoff & login to effect the permission changes.)
2. On Page 4, verify that the features shown in bold in the example below are enabled.
4. Use the change node-name ip command to assign the node name and IP address for
the Avaya SES at Site 1. In this case, SES and 50.1.1.50 are being used, respectively.
The node name SES will be used throughout the other configuration forms of Avaya
Communication Manager. In this example, procr and 50.1.1.10 are the name and IP
address assigned to the Avaya S8300 Media Server.
IP Codec Set
Codec Set: 1
7. On Page 2, the FAX Mode field must be set to t.38-standard to support the fax
machine. The Modem field must be set to off. The screen below shows the setting
used for the fax testing.
IP Codec Set
Mode Redundancy
FAX t.38-standard 0
Modem off 0
TDD/TTY US 3
Clear-channel n 0
Signaling Group: 1
Number of Members: 24
10. On Page 3:
Verify the Numbering Format field is set to public. This field specifies the
format of the calling party number sent to the far-end.
The default values may be retained for the other fields.
5 4 1 5
12. Create a route pattern that will use the SIP trunk that connects to Avaya SES. The
compliance test used Automatic Route Selection (ARS) to define route pattern 1 as the
route for all outbound calls. For more information on ARS see [1] and [2].
To create a route pattern, use the change route-pattern n command, where n is the
number of an unused route pattern. Enter a descriptive name for the Pattern Name
field. Set the Grp No field to the trunk group number created for the SIP trunk. Set
the Facility Restriction Level (FRL) field to a level that allows access to this trunk for
all users that require it. The value of 0 is the least restrictive level. The default values
may be retained for all other fields.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 3 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
Loc. Name Timezone Rule NPA ARS Attd Pre- Proxy Sel.
No. Offset FAC FAC fix Rte. Pat.
1: Main + 00:00 0 1
2:
3:
14. To map a DID number to a station at Site 1, use the change inc-call-handling-trmt
trunk-group n command, where n is the SIP trunk group number connected to Site 2
defined in Step 9. All inbound calls arrive at Site 1 via Site 2 which has PSTN access.
The example below shows two incoming 10-digit numbers being deleted and replaced
with the extension number of the desired station.
8. On Page 6, enter the group members. For each DS1 port to be added as a member of
the trunk group, enter the port number in the Port field and the corresponding signaling
group for that port in the Sig Grp field. The Code field is filled in automatically. In
the compliance test, each of the 23 bearer channels of the DS1 board added in Step 3
were added to this group. Only the first 15 members are shown below. The signaling
channel for each of these ports is the signaling channel added in Step 4.
10. The compliance testing used Automatic Route Selection (ARS) to define route pattern
12 as the route for all outbound calls to the PSTN on the Avaya DEFINITY Server R.
For more information on ARS see [1] and [2].
The example below shows the route pattern used in the compliance test for all
outbound traffic. The Pattern Name can be any descriptive name. The Grp No. is set
to the trunk-group number for the trunk to be used. The FRL field defines the facilty
restriction level for this route pattern. The value of 0 is the least restrictive. The Pfx
Mrk field is set to 1. The Prefix Mark sets the requirement for sending a prefix digit 1.
Setting the Pfx Mrk field to a 1 results in a 1 being prefixed to any 10-digit number.
An 11-digit number, presumably already with a 1, is left unchanged. Default values for
all other fields can be used.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 3 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
Step Description
1. Access the Avaya SES administration web interface by entering
http://<ip-addr>/admin as the URL in a Web browser, where <ip-addr> is the IP
address of the Avaya SES server.
Log in with the appropriate credentials and then select the Launch Administration
Web Interface link from the main page as shown below.
The Edit Host page shown below is accessible by clicking on the Hosts List link in
the left pane and then clicking on the Edit link under the Commands section of the
subsequent page that is displayed.
In the Host IP Address field, enter the IP address of the Avaya SES server.
Enter the DB Password that was specified during the system installation.
The default values for the other fields may be used.
Scroll down to the bottom of the page and click the Update button.
On the Add Media Server Interface page, enter the following information:
Enter a descriptive name in the Media Server Interface Name field (e.g.
S8300).
In the Host field, select the Avaya SES server from the pull-down menu that
will serve as the SIP proxy for this media server. Since there is only one Avaya
SES server in this configuration, the Host field is set to the host shown in Step
5.
Select TLS (Transport Link Security) for the SIP Trunk Link Type. TLS
provides encryption at the transport layer. TLS is the only link protocol that is
supported for communication between Avaya SES and Avaya Communication
Manager.
Enter the IP address of the Avaya S8300 Media Server in the SIP Trunk IP
Address field. In alternative configurations that use a C-LAN board, the SIP
Trunk IP Address would be the IP address of the C-LAN board.
The default values may be retained for all other fields.
After completing the Add Media Server Interface page, click the Add button.
In the case of the compliance test, the user portion contained the called party number.
All calls bound for the PSTN were routed to the Mediant 1000. Thus, the Host
Address Map was configured to match all calls dialed with an 11 digit number
beginning with a 1.
sip:$(user)@ip_addr:port;transport=protocol
The user part in the original request URI is inserted in place of the “$(user)”
string before the message is sent to the destination.
For the compliance test, the Mediant 1000 had IP address of 192.168.1.60.
Thus, the following contact value was used:
sip:$(user)@192.168.1.60:5060;transport=udp
11. A Media Server Address Map is required on Avaya SES to direct calls inbound to
Avaya Communication Manager from the Mediant 1000 in the same way that
outbound calls from Avaya Communication Manager required a Host Address Map.
In the case of the compliance test, calls from the PSTN beginning with 1732555 were
routed through the Avaya DEFINITY Server R and then from the Mediant 1000 to
Avaya Communication Manager at Site 1. The leading 1 was deleted by the Avaya
DEFINITY Server R at Site 2 before sending the call to the Mediant 1000. Thus, the
Media Server Address Map was configured to match all calls dialed with a 10 digit
number beginning with 732555.
13. After configuring the Media Server Address Map, the List Media Server Address
Map page appears as shown below. The first Media Server Contact is created
automatically and directs the calls to the IP address of the Avaya Media Server
(50.1.1.10) using port 5061 and TLS as the transport protocol. The user portion in the
original request URI is substituted for “$(user)”. For the compliance test, the Contact
field for the Media Server Address Map is displayed as:
sip:$(user)@50.1.1.10:5061;transport=tls
trustedhost –L
admin@denali> trustedhost -L
Third party trusted hosts.
Trusted Host | CCS Host Name | Comment
--------------------------+---------------------------+-------------------------
192.168.1.60 | 50.1.1.50 | Mediant1000
Step Description
1. The configuration of the Mediant 1000 is done via a Web browser. To access the
device, enter the IP address of the Mediant 1000 in the Address field of the browser.
2. The following pop-up window will appear. Login with the proper credentials.
Default values may be retained for all other fields. Scroll down to the bottom of the
page and click Submit (not shown).
Default values may be retained for all other fields. Scroll down to the bottom of the
page and click Submit (not shown).
The codec list used for the compliance test is shown in the example below. G.711U-
law was selected as the most preferred followed by G.729. For the G.729 codec, the
Silence Suppression field was set to Enable. With silence suppression enabled, the
Mediant 1000 uses the equivalent of a G.729AB codec. Default values were retained
for all other fields.
Click Submit.
Default values may be retained for all other fields. Click Submit.
Default values may be retained for all other fields. Click Submit.
The Trunk Group Table maps a particular trunk channel to a trunk group. In the From
Trunk and To Trunk columns, enter the starting and ending trunks to be assigned. In
the Channel(s) column, enter the range of channels on those trunks to be assigned. A
phone number may be entered in the Phone Number column or it may be left blank.
If a number is entered, this number will be used as the originating calling party if no
calling party information is received from the originating PSTN trunk. Each channel is
assigned a unique number starting with the value in the Phone Number column and
incrementing for each subsequent channel. If the Phone Number column is left blank,
the Mediant 1000 will use a default value for the originating calling party if no calling
party information is received from the originating PSTN trunk. In the Trunk Group
ID column, enter the trunk group that will contain these channels. The default value
may be used for the Profile ID column.
In the example below, the table entry assigns channels 1 – 24 of trunk 1 to Trunk
Group 1. A range of numbers arbitrary chosen to start at 44444 will be used for the
originating calling party number if no calling party information is received from the
originating PSTN trunk.
Click Submit.
Click Submit.
This table defines the mapping of PSTN trunk calls to an IP address that will process
the IP leg of the call. The Dest. Phone Prefix and Source Phone Prefix columns
define which calls are mapped to the IP address in the Dest. IP Address column.
In the example below, the table entry maps calls from any destination prefix, or any
source prefix to IP address 50.1.1.0 which is the address of the Avaya SES server. The
default values may be used for the Profile ID column.
Click Submit.
This table defines the mapping of IP calls to the trunk group created in Step 12. The
Dest. Phone Prefix, Source Phone Prefix and Source IP Address columns define
which calls are mapped to the trunk group in the Trunk Group ID column. In the
example below, the table entry maps calls from any destination prefix, or any source
prefix or any source IP address to trunk group 1.
Click Submit.
Default values may be retained for all other fields. Click Submit.
Scroll down to the bottom of the page and click Submit (not shown).
21. Navigate to ini Parameters in the left pane. In the Parameter Name field, select
ModemBypassPayloadType from the pull-down menu. Enter 0 in the Enter Value
field. Click on Apply New Value.
A value of 0 in the payload type indicates a G.711mu call. By default, the Mediant
1000 uses a proprietary value for the payload type of modem calls so they can be
distinguished from other G.711mu calls.
For telephony to IP calls, the Mediant 1000 can be configured to change the content of
the Contact header in the SIP messages it generates for these calls. This is done by
changing the value of the ISUSETOHEADERASCALLEDNUMBER parameter. By
default, this parameter is set to 0 and the originating PSTN calling number is used in
the Contact header. If this parameter is set to 1, then the gateway extension is used in
the Contact header. For interoperability, this parameter must be set to 1.
23. To save the configuration, return to the administrative web page and select
Maintenance in the left pane. Click the BURN button.
7. Verification Steps
The following steps may be used to verify the configuration:
• From the Avaya Communication Manager SAT, use the status signaling-group command to
verify that the SIP signaling group is in-service.
• From the Avaya Communication Manager SAT, use the status trunk-group command to
verify that the SIP trunk group is in-service.
• Verify that calls can be placed to/from the PSTN through the Mediant 1000.
8. Support
For technical support on the Mediant 1000, contact AudioCodes via the support link at
www.audiocodes.com.
Product documentation for AudioCodes Mediant 1000 VoIP Media Gateway products may be found
at http://www.audiocodes.com.
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).
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.
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
DeveloperConnection Program at devconnect@avaya.com.