Avaya CM SES - MP124 Configuration Guide

7/31/2019 Avaya CM SES - MP124 Configuration Guide

1/44

BS; Reviewed:

SPOC 9/20/2006

Solution & Interoperability Test Lab Application Notes

2006 Avaya Inc. All rights reserved.

1 of 44

AudioCodes-SIP.doc

Avaya Solution & Interoperability Test Lab

Application Notes for AudioCodes MP-124 Analog VoIP

Gateway with Avaya SIP Enablement Services and Avaya

Communication Manager Issue 1.0

Abstract

These Application Notes describe the configuration procedures required for the AudioCodes

MP-124 Analog VoIP gateway to successfully interoperate with Avaya SIP Enablement

Services and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124gateway provides voice technology to connect analog telephones, fax machines and modems

to the SIP servers and IP based PBX systems. In the reference test network simulating a hosted

telephony solution, this technology enabled the analog endpoints behind the AudioCodes MP-124 at the customer enterprise sites to register with Avaya SIP Enablement Services server at

the service provider data center, effectively making them SIP endpoints. These analog

endpoints were then able to utilize the telephony features of Avaya Communication Manager.


2/44

BS; Reviewed:

SPOC 9/20/2006



2 of 44

AudioCodes-SIP.doc

1. IntroductionThese Application Notes describe the configuration procedures required for the AudioCodesMP-124 Analog VoIP gateway to successfully interoperate with Avaya SIP Enablement Services

and Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 gatewayprovides voice technology to connect analog telephones, fax machines and modems to the SIPservers and IP based PBX systems. In the reference test network simulating a hosted telephony

solution, this technology enabled the audio endpoints behind the AudioCodes MP-124 at the

customer enterprise sites to register with Avaya SIP Enablement Services server at the serviceprovider data center, effectively making them SIP endpoints. These analog endpoints were then

able to utilize the telephony features of Avaya Communication Manager.

The MP-124 gateway incorporates up to 24 analog ports for connection to telephones, fax

machines and modems, supporting up to 24 simultaneous VoIP calls.

1.1. Reference Test Network

Figure 1 shows the reference configuration used for simulating a hosted telephony solution withtwo customer enterprise sites and a service provider data center.

The service provider data center site consists of an Avaya SIP Enablement Services (SES)server, an Avaya S8500 Media Server running Avaya Communication Manager and an Avaya

G650 Media Gateway. The access to the PSTN for all the enterprise customers was provided by

an ISDN PRI trunk from an Avaya G650 Media Gateway to the PSTN.

Each enterprise site consists of an AudioCodes MP-124 analog VoIP gateway, with the analog

ports connected to the analog telephones, a fax machine and a modem. The AudioCodes MP-124is connected to the service provider data center network over an IP network via an Avaya C363T

switch. Avaya 4600 Series H.323 IP telephones were also located at the enterprise sites to verify

calls to and from the analog endpoints behind MP-124 gateway. Avaya 4600 Series H.323 IPtelephones registered directly with Avaya Communication Manager.


3/44

BS; Reviewed:

SPOC 9/20/2006



3 of 44

AudioCodes-SIP.doc

Figure 1: Reference Test Configuration


4/44

BS; Reviewed:

SPOC 9/20/2006



4 of 44

AudioCodes-SIP.doc

2. Equipment and Software ValidatedThe following equipment and software/firmware were used for the test configuration provided.

Equipment Software/FirmwareAvaya S8500 Media Server Avaya Communication Manager 3.1.2

(R013x.01.0.632.1)

Avaya G650Media Gateway (Media GatewayProcessor)

25.23.0

Avaya SES 3.1(3.1.0.0-018.0)

Avaya 4600 Series IP (H.323) Telephones 2.3 (4620SW)

Analog Telephones -

Avaya 363T Switch 4.5.14

AudioCodes MP-124 analog VoIP gateway 4.80A.018.007

Fax machine -

PCs with modem Windows 2000

3. Configure Avaya Communication ManagerThis section describes the necessary configuration on Avaya Communication Manager to

interoperate with the SIP interface of the Avaya SES.

The following configuration of Avaya Communication Manager was performed using the

System Access Terminal (SAT). After completion of the configuration in this section, perform a

save translation command to make the changes permanent.

3.1. Configure SIP Trunking

This section describes the steps for configuring a SIP trunk group between Avaya

Communication Manager and Avaya SES. Avaya SES acts as a SIP proxy for the service

provider data center in Figure 1. Thus, the AudioCodes MP-124 analog SIP gateway will direct

all SIP traffic bound for the service provider data center to the Avaya SES located there. AvayaSES will then forward the inbound SIP traffic to Avaya Communication Manager. The path is

the same in the reverse direction with SIP signaling flowing from Avaya Communication

Manager to Avaya SES to the AudioCodes MP-124. As a result, the SIP trunk group configuredin this section carries all the SIP signaling and associated media streams bound to the service

provider data center from the AudioCodes MP-124 gateways at the customer enterprise sites andvice versa.

These steps were performed at the service provider data center in the configuration shown in

Figure 1.


5/44


6/44

BS; Reviewed:

SPOC 9/20/2006



6 of 44

AudioCodes-SIP.doc

Step Description

2. Use the change node-name ip command to assign the node name and IP address for

Avaya SES at the service provider data center. In this case, ses and 192.168.10.50 are

being used, respectively. The node name ses will be used throughout the otherconfiguration forms of Avaya Communication Manager. In this example, CLANand

192.168.10.43 are the name and IP address assigned to the CLAN IP interface onAvaya G650 Media Gateway.

change node-names ip Page 1 of 1

IP NODE NAMES

Name IP Address Name IP Address

CLAN 192.168.10 .43 . . .

MEDPRO 192.168.10 .42 . . .

default 0 .0 .0 .0 . . .

procr 192.168.10 .40 . . .

ses 192.168.10 .50 . . .

. . . . . .


7/44

BS; Reviewed:

SPOC 9/20/2006



7 of 44

AudioCodes-SIP.doc

Step Description

3. Use the change ip-network-regionn command, wheren is the number of the region

to be changed, to define the connectivity settings for all VoIP resources and IP

endpoints within the IP region. Select an IP network region that will contain the AvayaSES server. The association between this IP Network Region and the Avaya SES

server will be done on the Signaling Group form as shown in Step 5. In the case ofthe tested configuration, the same IP network region that contains the Avaya S8500

Media Server was selected to contain the Avaya SES server. By default, the MediaServer is in IP Network Region 1.

On the IP Network Region form: The Authoritative Domain field is configured to match the domain name

configured on Avaya SES. In this configuration, the domain name is test.com.

This name will appear in the From header of SIP messages originating fromthis IP region.

By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to

be sent directly between IP endpoints without using media resources in theAvaya G650 Media Gateway. This is true for both intra-region and inter-regionIP-IP Direct Audio. Shuffling can be further restricted at the trunk level on the

Signaling Group form.

The Codec Set is set to the number of the IP codec set to be used for callswithin this IP network region. In this configuration, this codec set will apply to

all calls to and from the AudioCodes MP-124 as well as any IP phone (H.323 or

SIP) within the enterprise. If different IP network regions are used for theAvaya S8500 Media Server and the Avaya SES server, then Page 3 of each IP

Network Region form must be used to specify the codec set for inter-region

communications.

The default values can be used for all other fields.

change ip-network-region 1 Page 1 of 19

IP NETWORK REGION

Region: 1

Location: Authoritative Domain: test.com

Name: CM

MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes

Codec Set: 1 Inter-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? n

UDP Port Max: 65531

DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y

Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS

Audio PHB Value: 46 Use Default Server Parameters? y

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 6

Audio 802.1p Priority: 6

Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS

H.323 IP ENDPOINTS RSVP Enabled? n

H.323 Link Bounce Recovery? y

Idle Traffic Interval (sec): 20

Keep-Alive Interval (sec): 5

Keep-Alive Count: 5


8/44

BS; Reviewed:

SPOC 9/20/2006



8 of 44

AudioCodes-SIP.doc

Step Description

4. Use the change ip-codec-setn command, wheren is the codec set value specified in

Step 3,to enter the supported audio codecs for calls routed to Avaya SES and the

AudioCodes MP-124. Multiple codecs can be listed in priority order to allow thecodec to be negotiated during call establishment. The list should include the codecs the

enterprise wishes to support within the normal trade-off of bandwidth versus voicequality. It must also include the preferred codec configured on the AudioCodes MP-

124. The example below shows the values used in the tested configuration.

change ip-codec-set 1 Page 1 of 2

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.711MU n 2 20

2: G.729AB n 2 20

3:

4:

Page forward to Page 2. If the T.38 faxing is desired, enter t.38-standardin the FAX

field. For data modem calls, retain the default value ofofffor the Modem field.

change ip-codec-set 1 Page 2 of 2

IP Codec Set

Allow Direct-IP Multimedia? n

Mode Redundancy

FAX t.38-standard 0

Modem off 0

TDD/TTY US 3Clear-channel n 0


9/44

BS; Reviewed:

SPOC 9/20/2006



9 of 44

AudioCodes-SIP.doc

Step Description

5. Use the add signaling groupn command, wheren is the number of an unused

signaling group, to create the SIP signaling group as follows:

Set the Group Type field to sip.

The Transport Method field will default to tls (Transport Layer Security).

TLS is the only link protocol that is supported for communication betweenAvaya SES and Avaya Communication Manager.

Specify the node name CLANon Avaya S8500 Media Server and the nodename ses for Avaya SES Server as the two ends of the signaling group in the

Near-end Node Name and the Far-end Node Name fields, respectively.

These field values are taken from the IP Node Names form shown in Step 2.In alternative configurations, which do not use a CLAN board, the near (local)

end of the SIP signaling group will be theprocrinstead of the CLAN board.

Ensure that the recommended TLS port value of5061 is configured in the

Near-end Listen Port and the Far-end Listen Port fields.

In the Far-end Network Region field,enter the IP Network Region value

assigned in the IP Network Region form in Step 3. This defines which IPnetwork region contains the Avaya SES server. If the Far-end Network

Region field is different from the near-end network region, the preferred codec

will be selected from the IP codec set assigned for the inter-region connectivity

for the pair of network regions. Enter the domain name of Avaya SES in the Far-end Domain field. In this

configuration, the domain name is test.com as configured in Step 3. This

domain is specified in the Uniform Resource Identifier (URI) of the SIP Toheader in the INVITE message.

The Direct IP-IP Audio Connections field is set toy to enable the direct

audio path (shuffling) between the two IP endpoints.

The DTMF over IP field must be set to the default value ofrtp-payloadfor aSIP trunk. This value enables Avaya Communication Manager to send DTMF

transmissions using RFC 2833.

The default values for the other fields may be used.change signaling-group 1 Page 1 of 1

SIGNALING GROUP

Group Number: 1 Group Type: sip

Transport Method: tls

Near-end Node Name: CLAN Far-end Node Name: ses

Near-end Listen Port: 5061 Far-end Listen Port: 5061

Far-end Network Region: 1

Far-end Domain: test.com

Bypass If IP Threshold Exceeded? y

DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y

IP Audio Hairpinning? y


10/44

BS; Reviewed:

SPOC 9/20/2006



10 of 44

AudioCodes-SIP.doc

Step Description

6. Add a SIP trunk group by using the add trunk-groupn command, wheren is the

number of an unused trunk group. For the compliance test, trunk group number 1 was

chosen.

On Page 1, set the fields to the following values: Set the Group Type field to sip.

Choose a descriptive Group Name. Specify an available trunk access code (TAC) that is consistent with the

existing dial plan.

Set the Service Type field to tie. Specify the signaling group associated with this trunk group in the Signaling

Group field as previously specified in Step 5.

Specify the Number of Members supported by this SIP trunk group. Asmentioned earlier, each SIP call between two SIP destinations (whether internal

or external) requires two SIP trunks for the duration of the call. Thus, a call

between two SIP extensions, e.g., two telephones behind the AudioCodes MP-124, will use two SIP trunks. A call between a SIP extension and the PSTNwill only use one SIP trunk.

The default values may be retained for the other fields.

add trunk-group 1 Page 1 of 21

TRUNK GROUP

Group Number: 1 Group Type: sip CDR Reports: y

Group Name: siproute COR: 0 TN: 1 TAC: 701

Direction: two-way Outgoing Display? n

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Signaling Group: 1Number of Members: 100

7. On Page 3:

Set the Numbering Format field topublic. 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.

add trunk-group 1 Page 3 of 21

TRUNK FEATURES

ACA Assignment? n Measured: none

Maintenance Tests? y

Numbering Format: public

Prepend '+' to Calling Number? n

Replace Unavailable Numbers? n


11/44

BS; Reviewed:

SPOC 9/20/2006



11 of 44

AudioCodes-SIP.doc

Step Description

8. Configure a route pattern that is routing to its outbound SIP proxy server (Avaya SES

server). This will be assigned on the locations screen. Refer to Step 9.

The example belowshows the route pattern used in the reference test configuration.

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 facility restriction

level for this route pattern. The value of0 is the least restrictive. Default values for allother fields can be used.

change route-pattern 1 Page 1 of 3

Pattern Number: 1 Pattern Name: siproute

SCCAN? n Secure SIP? n

Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC

No Mrk Lmt List Del Digits QSIG

Dgts Intw

1: 1 0 n user

2: n user

3: n user

4: n user

5: n user6: n user

9. Use the change locations command to assign the route pattern to the location. Thisscreen allows for each location to point to the route pattern that is routing to its

outbound SIP proxy server. This correlation is required by features and services such

as Transfer and URI Dialing. Only one location created by default, known as Main,

exists for the service provider data center and the customer sites. Enter the routepattern number from the previous step in the Proxy Sel. Rte Pat. field. The defaultvalues may be retained for all other fields.

change locations Page 1 of 16

LOCATIONS

ARS Prefix 1 Required For 10-Digit NANP Calls? y

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:

4: :


12/44

BS; Reviewed:

SPOC 9/20/2006



12 of 44

AudioCodes-SIP.doc

3.2. Configure SIP Stations

The analog endpoints (e.g. analog telephones, fax machines and modems) behind the

AudioCodes MP-124 are considered SIP stations from the point of view of Avaya

Communication Manager. This section describes the steps for configuring SIP stations. Supportfor SIP stations that use SIP trunks allows any fully compliant SIP telephone to interoperate with

Avaya telephones. This means that any SIP telephone can:

Dial and be dialed as an extension in the enterprise dial plan. Put calls on hold and participate in transfers and conference calls.

SIP stations that are administered as Outboard Proxy SIP stations, also known as off-PBX station(OPS) stations, support most Extended Access Features, such as call park, call pick-up, and

priority calls. To activate these features, either dial Feature Access Codes or dial special

extensions, that is, Feature Name Extensions. For more details, seeAvaya Extension to Cellular

Users Guide, Issue 6, DocID 210-100-700, and theAvaya Extension to Cellular and OPSInstallation and Administration Guide, Issue 8, DocID 210-100-500. The configuration for

Feature Access Codes and Feature Name Extensions is not in the scope of this document.

These steps were performed using the SAT at the service provider data center in the

configuration shown in Figure 1 to add the OPS stations.


13/44

BS; Reviewed:

SPOC 9/20/2006



13 of 44

AudioCodes-SIP.doc

Step Description

1. Use the displaysystem-parameters customer-options command to verify that

Maximum Off-PBX Telephones OPS has been set to a value that will accommodatethe number of telephones to be supported.

display system-parameters customer-options Page 1 of 10

OPTIONAL FEATURES

G3 Version: V13

Location: 1 RFA System ID (SID): 1

Platform: 12 RFA Module ID (MID): 1

USED

Platform Maximum Ports: 3200 231

Maximum Stations: 2400 21

Maximum XMOBILE Stations: 0 0

Maximum Off-PBX Telephones - EC500: 2400 0

Maximum Off-PBX Telephones - OPS: 2400 16

Maximum Off-PBX Telephones - SCCAN: 0 0

(NOTE: You must logoff & login to effect the permission changes.)

2. Use the add stationx command, wherex is an unused extension to create a station

extension as follows:

Leave the default value for Type field to 6408D+.

Set the Port field toX. The valueXimplies that this station is beingadministered without hardware such as a port on an Avaya analog or a digital

circuit pack. Set the Name field to any alphna-numeric name.

add station 2346241 Page 1 of 4

STATION

Extension: 2346241 Lock Messages? n BCC: 0

Type: 6408D+ Security Code: TN: 1

Port: X Coverage Path 1: COR: 1

Name: CM CUST A Analog 2 Coverage Path 2: COS: 1

Hunt-to Station:

STATION OPTIONS

Loss Group: 2 Personalized Ringing Pattern: 1

Data Module? n Message Lamp Ext: 2346241

Speakerphone: 2-way Mute Button Enabled? y

Display Language: english

Media Complex Ext:

IP SoftPhone? n


14/44

BS; Reviewed:

SPOC 9/20/2006



14 of 44

AudioCodes-SIP.doc

Step Description

3. Page forward to Page 3 and assign the buttons under BUTTON ASSIGNMENTS as

follows:

Make sure that at least two call-appr buttons are administered.

Assign no-hld-cnf button to enable the station to activate the Conference on

Answerfeature. Assign auto-cback button to enable the station to activate theAutomatic

CallBackfeature.Note that these are virtual button assignments that are used internally by Avaya

Communication Manager to implement the associated features.

add station 2346241 Page 3 of 4

STATION

SITE DATA

Room: Headset? n

Jack: Speaker? n

Cable: Mounting: d

Floor: Cord Length: 0

Building: Set Color:

ABBREVIATED DIALING

List1: group 1 List2: List3:

BUTTON ASSIGNMENTS

1: call-appr 5: auto-cback

2: call-appr 6:

3: 7:

4: no-hld-cnf 8:


15/44

BS; Reviewed:

SPOC 9/20/2006



15 of 44

AudioCodes-SIP.doc

Step Description

4. Use the add off-pbx-telephone station-mapping to associate an administered

extension with an external phone number as follows:

Set the Station Extension field to the extension administered in Step 2.

The Application field indicates the type of off-pbx application associated with

an internal phone extension. Set this field to OPSto indicate SIP EnablementServices application.

In the Phone Number field, enter the phone number of the off-pbx telephone.The off-pbx telephone is the analog telephone behind the AudioCodes MP-124.

Refer to Section 5 Steps 10-11.

Set the Trunk Selection field to 1, same as the trunk group configured inSection 3.1, Step 6.

The Configuration Set field indicates the configuration set number that defines

the desired call treatment options. Set this to 1. See Step 5.

add off-pbx-telephone station-mapping Page 1 of 2

STATIONS WITH OFF-PBX TELEPHONE INTEGRATION

Station Application Dial Phone Number Trunk Configuration

Extension Prefix Selection Set

2346241 OPS - 2346241 1 1

-

-

5. Use the change off-pbx-telephone configuration-set ncommand, wheren is a

configuration set number to assign the call treatment options as follows:

Set the Fast Connect on Origination field toy to allow Avaya Communication

Manager to do some additional processing prior to connecting the call. Leave other fields to their default values.

change off-pbx-telephone configuration-set 1 Page 1 of 1

CONFIGURATION SET: 1

Configuration Set Description: 1

Calling Number Style: network

CDR for Origination: phone-number

CDR for Calls to EC500 Destination? y

Fast Connect on Origination? y

Post Connect Dialing Options: dtmf

Cellular Voice Mail Detection: none

Barge-in Tone? nCalling Number Verification? y

Identity When Bridging: principal

6. Repeat Steps 1-4 for each analog endpoint behind the AudioCodes MP-124.


16/44

BS; Reviewed:

SPOC 9/20/2006



16 of 44

AudioCodes-SIP.doc

4. Configure Avaya SESThis section covers the configuration of Avaya SES. Avaya SES is configured via an Internetbrowser using the administration web interface. It is assumed that Avaya SES software and the

license file have already been installed on the server. During the software installation, theinstallation script is run from the Linux shell of the server to specify the IP network properties ofthe server along with other parameters. For additional information on these installation tasks,

refer to [3].

Step Description

1. Access the Avaya SES administration web interface, by entering

http:///admin as the URL in an Internet browser, where is the IP

address of Avaya SES.

Log in with the appropriate credentials and then select the Launch Administration

Web Interface link from the main page as shown below.


17/44

BS; Reviewed:

SPOC 9/20/2006



17 of 44

AudioCodes-SIP.doc

Step Description

2. The Avaya SES Administration Home Page shown below will be displayed.


18/44

BS; Reviewed:

SPOC 9/20/2006



18 of 44

AudioCodes-SIP.doc

Step Description

3. After making changes within Avaya SES, it is necessary to commit the database

changes using the Update link that appears when changes are pending. Perform this

step by clicking on the Update link found in the bottom of the blue navigation bar onthe left side of any of the Avaya SES administration pages as shown below. It is

recommended that this be done after making each set of changes described in thefollowing steps.


19/44

BS; Reviewed:

SPOC 9/20/2006



19 of 44

AudioCodes-SIP.doc

Step Description

4. From the left pane of the administration web interface, expand the Server

Configuration option and select System Properties. This page displays the software

version in the SES version field and the network properties entered via the installationscript during the installation process.

On the Edit System Properties page:

Enter the SIP Domain name assigned to Avaya SES. This must match theAuthoritative Domain field configured on Avaya Communication Manager

shown inSection 3.1 Steps 3 and 5.

Enter the License Host field. This is the host name, the fully qualified domainname, or the IP address of the SIP proxy server that is running the WebLM

application and has the associated license file installed. In this case, the

WebLM server is same as the Avaya SES server.

After configuring the Edit System Properties page, click the Update button

(not shown).


20/44

BS; Reviewed:

SPOC 9/20/2006



20 of 44

AudioCodes-SIP.doc

Step Description

5. After setting up the domain on the Edit System Properties page, create a host

computer entry for Avaya SES. The following example shows the Edit Host page

since the host had already been added to the system.

The Edit Host page shown below is accessible by clicking on the Hosts link in the leftpane and then clicking on the edit option under the Commands section of the

subsequent page that is displayed. Enter the Logical IP or Logical Name of this server in the Host IP Address,

the IP address of Avaya SES.

Enter the DB Password that was specified while running the installation scriptduring the system installation.

Verify the Host Type field. Since only one Avaya SES server exists in the test

configuration, the Avaya SES server provides the functionality of both a homeand edge server. Thus, the Host Type is configured as home/edge.

The default values for the other fields may be used.

Click the Update button.


21/44

BS; Reviewed:

SPOC 9/20/2006



21 of 44

AudioCodes-SIP.doc

Step Description

6. From the left pane of the administration web interface, expand the Media Server

option and select Add to add the Avaya Media Server to the list of media servers

known to Avaya SES. Adding the media server will create the Avaya SES side of theSIP trunk previously created in Avaya Communication Manager.

On the Add Media ServerInterface page, enter the following information:

A descriptive name in the Media Server Interface Name field (e.g.,s8500clan).

In the Host field, select the IP address of the Avaya SES server from the pull-

down menu that will serve as the SIP proxy for this media server. Since there isonly 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 Link Type. TLS providesencryption 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 C-LAN in the SIP Trunk IP Address field. Inalternative configurations, that do not use a C-LAN board, the SIP Trunk IP

Address would be the IP address of the Avaya Media Server.

The default values may be retained for all other fields.

After completing the Add Media ServerInterface page, click the Add button.


22/44

BS; Reviewed:

SPOC 9/20/2006



22 of 44

AudioCodes-SIP.doc

Step Description

7. Inbound SIP calls arriving at Avaya SES are routed to the appropriate Avaya

Communication Manager for termination services. This routing is specified in a Media

Server Address Map configured on Avaya SES.

This routing compares the Uniform Resource Identifier (URI) of an incoming INVITEmessage to the pattern configured in the Media Server Address Map , and if there is a

match, the call is routed to the designated Avaya Communication Manager. The URIusually takes the form ofsip:user@domain, where domain can be a domain name or an

IP address. Patterns must be specific enough to uniquely route incoming calls to the

proper destination if there are multiple Avaya Media Servers supported by the AvayaSES server.

In this test configuration, some of the incoming calls from customer sites require aMedia Server Address Map entry.

For example, in the test configuration, 3-digit numbers beginning with digit 8 wereconfigured as the Avaya Communication Manager Feature Access Codes. Fromcustomer sites, the userportion of the SIP URI will contain the Feature Access Code.

An example of a SIP URI in an INVITE message received from a customer site wouldbe:

sip: [email protected].

In this example, the user portion is the Feature Access Code (FAC) 841.

Thus, the media server address map strategy was to define pattern matches for 8xx inthe URI, where x is any digit between 0 and 9, and have Avaya SES forward the

messages that match to the appropriate media server.

To configure a Media Server Address Map:

Expand the Media Servers option in the left pane of the administration web

interface and select List. This will display the List Media Servers page. Click on the Map link associated with the appropriate media server to display

the List Media Server Address Map page.

Click on the Add Map In New Group link. The page shown below isdisplayed. The Host field displays the name of the media server to which this

map applies.

Enter a descriptive name in the Name field.

Enter the regular expression to be used for the pattern matching in the Patternfield. The example below shows the pattern specification for Feature Access

Codes ^sip:8[0-9]{2}. This expression will match any SIP URI that

begins with the text string sip:8followed by a 2-digit string. Based on the

value of the Host field, these SIP calls will then be routed to host S8500clan.


23/44

BS; Reviewed:

SPOC 9/20/2006



23 of 44

AudioCodes-SIP.doc

Step Description

Appendix A provides a detailed description of the syntax for address map

patterns.

Click the Add button once the form is completed.


24/44

BS; Reviewed:

SPOC 9/20/2006



24 of 44

AudioCodes-SIP.doc

Step Description

8. After configuring the media server address map, the List Media Server Address Map

page appears as shown below, showing an entry for 2xx Address Map.

In addition to the dialed phone numbers, the Media Server Address Map may also beneeded for other user dialable digit strings such as Abbreviated Dial number or a

Feature Access Code administered on Avaya Communication Manager. The list shows

other such maps (e.g., AbbDialing for abbreviated dialing) the patterns are not shown.

After the first Media Server Address Map is added, the Media Server Contact is createdautomatically. For the Media Server Address Map that was added, the following

contact was created and displayed in the Contact field:

sip:$(user)@192.168.10.43:5061;transport=tls

The contact specifies the IP address of the CLAN board on Avaya S8500 Media Serverand the transport protocol used to send SIP signaling messages. The user in the

original request URI is substituted for $(user).


25/44

BS; Reviewed:

SPOC 9/20/2006



25 of 44

AudioCodes-SIP.doc

Step Description

9. From the left pane of the administration web interface, expand the User option and

select Add to add a user and an extension for an analog endpoint behind the

AudioCodes MP-124.

Specify a Primary Handle and User ID. A handle specifies a user on Avaya

SES. The User ID is used to authenticate a user on Avaya SES. The User IDmust be same as the User ID configured on the AudioCodes MP-124 for an

analog telephone. Refer Section 5, Step 11. Specify a Password and Confirm Password. The Password along with the

User ID is used to authenticate a user on Avaya SES. The Password must be

same as the Password configured on the AudioCodes MP-124 for an analogtelephone. See Section 5, Step 11.

The Host field is pre-populated with that Avaya SES Host IP Address since

there is only one Avaya SES in the test configuration.

Enter any descriptive First Name and Last Name.

Select Add Media Server Extension to add an OPS extension for the user.

Refer Step 11. Click the Add button.


26/44

BS; Reviewed:

SPOC 9/20/2006



26 of 44

AudioCodes-SIP.doc

Step Description

10. The following Continue page appears, confirming that the user was added

successfully.

ClickContinue.

11.

The Add Media Server Extension page will appear as shown below. Enter the Extension, same as the OPS extension configured in Avaya

Communication Manager. Refer to Section 3.2, Step 2.

Select the appropriate Media Server configured in Step 6 .

ClickAdd.

12. Repeat Steps 9-11 to add the users and their associated extensions for additional analog

endpoints behind the AudioCodes MP-124.


27/44

BS; Reviewed:

SPOC 9/20/2006



27 of 44

AudioCodes-SIP.doc

Step Description

13. To view the configured users, from the left pane of the administration web interface,

expand the Users option and select List.

The List Users page will appear as shown below.

14. To view the configured extensions, from the left pane of the administration web

interface, expand the Media Server Extensions option and select List.

The List Media Server Extensions page will appear as shown below.


28/44

BS; Reviewed:

SPOC 9/20/2006



28 of 44

AudioCodes-SIP.doc

5. Configure AudioCodes MP-124 VoIP GatewayAudioCodes-MP124 is configured via an Internet browser using the Embedded Web Serverinterface. The following steps describe how to configure:

IP address of the AudioCodes MP-124 and associating SIP proxy (Avaya SES). SIP proxy (Avaya SES) & registration.

General Parameters

Coders DTMF and Dialing

Advanced Control Parameters

Supplementary Services such as Hold and Transfer.

Endpoint Phone Numbers.

Endpoint Authentication.

Enable Caller ID

T.38 Fax

Step Description

1. Access the AudioCodes MP-124 Embedded Web Server interface, by enteringhttp:/// as the URL in an Internet browser, where is the default IP

address of the AudioCodes MP-124. Refer to [6] for the default IP address.


29/44

BS; Reviewed:

SPOC 9/20/2006



29 of 44

AudioCodes-SIP.doc

Step Description

2. IP address and SIP Proxy (Avaya SES)The web interface screen appears as shown below with Quick Setup selected.

To assign an IP address for the AudioCodes-MP124, under IP Configuration enter the

IP Address, Subnet Mask, and Default Gateway IP Address

To assign basic SIP parameters, under SIP Parameters assign the parameters as shownbelow:

Set Working with Proxy field to Yes to enable the interworking of the

AudioCodes MP-124 with Avaya SES proxy server. Enter the IP address of the Avaya SES in the Gateway Name, Proxy IP

Address and Proxy Name fields.

Select Enable Registration toEnable. The AudioCodes will send REGISTERrequest to Avaya SES according to the Authentication Mode parameter. Refer

to Step 4 for Authentication Mode.

ClickReset button and clickOK in the prompt. The AudioCodes MP-124applies the changes and restarts.

After the AudioCodes MP-124 restarts, access the AudioCodes MP-124 Embedded

Web Server interface, by entering http:/// as the URL in an Internet browser,where is the new IP address of the AudioCodes MP-124 configured in this

step.


30/44

BS; Reviewed:

SPOC 9/20/2006



30 of 44

AudioCodes-SIP.doc

Step Description

3. From the left pane of the web interface, click the Protocol Management option. The

right pane displays configuration options as shown below.

4. SIP Proxy & RegistrationFrom the right pane of the web interface, click the Protocol Definition Proxy &

Registration option as shown below.


31/44

BS; Reviewed:

SPOC 9/20/2006



31 of 44

AudioCodes-SIP.doc

Step Description

The basic SIP proxy and registration parameters were configured in Step 2. Configure

additional SIP parameters as shown below.

Set the Registration Time to 3600 to specify that the registration for the analogendpoints behind the AudioCodes MP-124 is valid for one hour.

Select Send All Invite to Proxy field to Yes so that all the Invite requestsincluding those generated as a result of transfer or redirect are sent to Avaya

SES. Enter any alphanumeric string in the Cnonce field. This field is used for the

mutual authentication of the server and client.

Select the Authentication Mode as Per Endpoint. This implies that theregistration and authentication of each audio endpoint behind the AudioCodes

MP-124 with Avaya SES is done separately.


After completing this screen, click the Submit button.


32/44

BS; Reviewed:

SPOC 9/20/2006



32 of 44

AudioCodes-SIP.doc

Step Description

5. General ParametersFrom the right pane of the web interface, click the Protocol Definition GeneralParameters option. Configure general SIP parameters as shown below.

Select the Enable Early Media field asEnable.If enabled, the AudioCodes

MP-124 gateway sends 183 Session Progress response with Session DescriptionProtocol parameters(instead of 180 Ringing), allowing the media stream to be

set up prior to the answering of the call. Select Fax Signaling Method field as T.38 Relay if the T.38 fax transmission is

desired.

Select the Use user=phone in SIP URL field asNo. The default values may be retained for all other fields.

After completing this screen, click the Submit button (not shown here).

Refer to Step 14 for other T.38 fax settings.


33/44

BS; Reviewed:

SPOC 9/20/2006



33 of 44

AudioCodes-SIP.doc

Step Description

6. CodersFrom the right pane of the web interface, click the Protocol Definition Coders

option. From the Coders screen, configure the first to fifth preferred coders (and theirattributes) for the AudioCodes MP-124 gateway. The first coder is the highest priority

coder and is used by the gateway whenever possible. If the far end gateway (on AvayaCommunication Manager) cannot use the coder assigned as the first coder, the gateway

attempts to use the next coder and so forth.

Configure Coder parameters as shown below.

From the Coder Name drop-down list, select the appropriate coder. Refer toSection 3.1, Step 4 for the coder selection and preferred order in Avaya

Communication Manager.

For G.729 coder, enable the silence suppression by selectingEnable from the

Silence Suppression drop down list. With silence suppression enabled, the

gateway includes the string annexb=yes in the SDP of the relevant SIP

messages, an equivalent of G.729AB. Refer to Section 3.1, Step 4 for the codersetting in Avaya Communication Manager. The default values may be retained for all other fields.



34/44

BS; Reviewed:

SPOC 9/20/2006



34 of 44

AudioCodes-SIP.doc

Step Description

7. DTMF and DialingFrom the right pane of the web interface, click the Protocol Definition DTMF &Dialing option. Use this screen to configure parameters that are associated with DTMFand dialing. Configure the DTMF and Dialing parameters as shown below.

Enter the maximum number of digits that can be dialed (e.g., 18) in the MaxDigits in Phone Num field.

From the 1st

Tx DTMF Option drop-down list, selectRFC 2833 as thepreferred transmit DTMF negotiation method.

For the RFCC 2833 Payload Type field, enter 101.

Assign the digit map pattern in the Digit Mapping Rules field. If the digitstring (dialed number) has matched one of the patterns in the digit map, the

AudioCodes MP-124 gateway stops collecting digits and starts to establish a

call with the collected number. The digit map pattern contains up to 52 optionsseparated by a vertical bar (|). The maximum length of the entire digit pattern is

limited to 152 characters.

For example: 2xxxxxx|1xx|91xxxxxxxxxx|8xx|3xx was used in the testconfiguration, where 2xxxxxx was used for dialing telephone extensions, 1xx

was used for speed dialing (Abbreviated Dialing feature in Avaya

Communication Manager), 91xxxxxxxxxx for PSTN calls, 8xx for FeatureAccess Codes in Avaya Communication Manager and 3xx for Feature Name

Extensions in Avaya Communication Manager.

The default values may be retained for all other fields. After completing thisscreen, click the Submit button.


35/44

BS; Reviewed:

SPOC 9/20/2006



35 of 44

AudioCodes-SIP.doc

Step Description

8. Advanced Control ParametersFrom the right pane of the web interface, click the Advanced Parameters GeneralParameters option.

Configure the advanced control parameters as shown below: From the Enable Current Disconnect drop down list (under Disconnect and

Answer Supervision), selectEnable. The AudioCodes MP-124 gatewaygenerates a Current Disconnect Pulse to an analog endpoint behind the

AudioCodes MP-124 gateway when the call on this endpoint is released from

the Avaya SES. Set Broken Connection Timeout to desired amount of time an RTP packet

isnt received, after which the call is disconnected.

Set the Max Number of Active Calls to the maximum number of channelsavailable on the gateway. For the tested configuration, it was set to 24.




36/44

BS; Reviewed:

SPOC 9/20/2006



36 of 44

AudioCodes-SIP.doc

Step Description

9. Supplementary ServicesFrom the right pane of the web interface, click the Advanced Parameters Supplementary Services option.

Enable the services like hold and transfer as shown below: From the Enable Caller ID drop down list selectEnable. When enabled, the

calling number and name are sent to the analog endpoints behind theAudioCodes MP-124 gateway.

The default values may be retained for all other fields. Hold, Transfer, Call

Forwarding and Call Waiting are enabled by default. After completing this screen, click the Submit button.


37/44

BS; Reviewed:

SPOC 9/20/2006



37 of 44

AudioCodes-SIP.doc

Step Description

10. Endpoint Phone NumbersFrom the right pane of the web interface, click the Endpoint Phone Numbers option.

Enable and assign telephone numbers for the VoIP gateway ports on the AudioCodes

MP-124, to which the analog endpoints are connected.

Channel is a port number on the back of the AudioCodes MP-124. To enable a

channel, enter a port number. In the Phone Number field, enter the telephone number assigned to that

channel. For example, the telephone number 2346241 is assigned to channel 1.

Refer to Section 3.2 and Section 4 Step 9-14 for how to configure relatedtelephone extensions in Avaya Communication Manager and Avaya SES.



38/44

BS; Reviewed:

SPOC 9/20/2006



38 of 44

AudioCodes-SIP.doc

Step Description

11. Endpoint AuthenticationFrom the right pane of the web interface, click the Endpoint Settings Authentication option.

The Authentication Table defines a username and password combination forauthentication for each AudioCodes MP-124 port as shown below:

Enter the User Name and Password for each port. This is used for registrationand authentication with Avaya SES. Make sure that these values are the same

as configured using Section 4, Step 9.



39/44

BS; Reviewed:

SPOC 9/20/2006



39 of 44

AudioCodes-SIP.doc

Step Description

12. Enable Caller IDFrom the right pane of the web interface, click the Endpoint Settings GenerateCaller ID to Tel option.

The Generate Caller ID to Tel table is used to enable or disable (per port) the Caller IDgeneration and detection as shown below.

From the Caller ID drop down list for each port, selectEnable orDisable. Inthe tested configuration the Caller ID was enabled for the Ports 1-3. When

enabled, an analog telephone with a caller ID display capability displays the

calling number and name on receiving calls beginning with the second ringcycle. The Caller ID type defaults toBellCore - see Step 9. If set to other call

types, the caller ID display and behavior may be different.



13. From the left pane of the web interface, click the Advanced Configuration option.The right pane displays configuration options as shown below.


40/44

BS; Reviewed:

SPOC 9/20/2006



40 of 44

AudioCodes-SIP.doc

Step Description

14. T.38 FaxFrom the right pane of the web interface, click the Media Settings Fax/Modem/CID Settings option and configure the Fax option as shown below:

From the Fax Transport Mode drop down list, select T.38 Relay if the T.38

faxing is desired. Use the default values for all other fields.


Refer to Step 5 for other T.38 fax settings.

6. Verification StepsThis section provides verification steps that may be performed to verify that the solutiondescribed in these Application Notes is configured properly.

Verify that the SIP trunk group is in-service. To do this, use the status trunkncommand, wheren is the number of the trunk group to be verified.

Verify that the SIP signaling group is in-service. To do this, use the status signaling-

groupn command, wheren is the number of the signaling group to be verified.


41/44

BS; Reviewed:

SPOC 9/20/2006



41 of 44

AudioCodes-SIP.doc

Verify that an analog telephone behind the AudioCodes MP-124 can register with AvayaSES. To do this, from the left pane of the administration web interface of Avaya SES,

expand the User option and select Registered Users.

Verify that a call can be placed between two analog telephones behind the AudioCodesMP-124.

Verify that a call can be placed between an analog telephone behind the AudioCodes MP-124 and a telephone in the PSTN.

Verify that a call can be placed between an analog telephone behind the AudioCodes MP-

124 and an Avaya 4600 Series H.323 IP telephone.

Verify that a fax can be transmitted between a fax machine behind the AudioCodes MP-

124 and a fax machine in the PSTN. Verify that a data modem call can be placed between a data modem behind the

AudioCodes MP-124 and a data modem in the PSTN.

Verify that a T.38 fax can be transmitted between a fax machine behind the AudioCodesMP-124 and a fax machine in the PSTN.

Verify that the conference, transfer, and hold features can be successfully activated from

the analog telephones behind the AudioCodes MP-124. Verify that the DTMF digits can be transmitted over an established call.

Verify a sample of other Avaya Communication Manager features (e.g., hunt group, call

pick up group, call forward, send all calls, and call waiting) that involve the analog

telephones behind the AudioCodes MP-124.

The following exceptions were observed during the testing.

The analog telephones behind the AudioCodes MP-124 cannot activate conference. For T.38 Fax, an unofficial patch was required on Avaya Communication Manager to

make it work. The fix will be available in the next Generally Available release of AvayaCommunication Manager.

The following Avaya Communication Manager features worked, however the audible

alerting was not exactly as expected:o Call forward unconditional: the calls are forwarded but no alerting on the

forwarded (analog) station.

o Send All Calls (SAC): SACd station hears one full ring instead of an abbreviatedring or tone.

7. SupportTechnical support for the AudioCodes MP-124 can be obtained from AudioCodes. See the

Support link at www.audiocodes.com for contact information.

8. ConclusionThe analog telephones, modems and fax machines behind the AudioCodes MP-124 can

successfully register to Avaya SIP Enablement Services (SES) and utilize telephony features ofAvaya Communication Manager.


42/44

BS; Reviewed:

SPOC 9/20/2006



42 of 44

AudioCodes-SIP.doc

9. Additional ReferencesThe following Avaya product documentation can be found at http://support.avaya.com.

[1] Feature Description and Implementation For Avaya Communication Manager, Issue 4.0,

February 2006, Document Number 555-245-205.

[2] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document

Number 03-300509.[3] Installing and Administering SIP Enablement Services R3.1, Issue 1.5, February 2006

Document Number 03-600768.[4] SIP Support in Release 3.1 of Avaya Communication Manager Running on the Avaya S8300,S8500, S8500B, S8700, and S8710 Media Server, February 2006, Issue 6, Document Number555-245-206.

[5] 4600 Series IP Telephone Release 2.4 LAN Administrator Guide, April 2006, Issue 2.3,

Document Number 555-233-507.

The following AudioCodes MP-124 product documentation is available from AudioCodes. Visit

http://www.audiocodes.com for company and product information.

[6] AudioCoded MediaPack MP-124 and MP-11x Users Manual 4.8.


43/44

BS; Reviewed:

SPOC 9/20/2006



43 of 44

AudioCodes-SIP.doc

APPENDIX A: Specifying Pattern Strings in Address MapsThe syntax for the pattern matching used within 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, andspecial metacharacters, which may represent items like quantity, location or types of

character(s).

In the pattern matching string used in 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 the 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 Curley brackets containing an integer n indicate that the preceding character

must be matched exactly n time. 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 willmatch.

A typical INVITE request below uses the shaded portion to illustrate the matching pattern.

INVITE sip:[email protected]:5060;transport=udp SIP/2.0


44/44

2006 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 issubject to change without notice. The configurations, technical data, and recommendations

provided in these Application Notes are believed to be accurate and dependable, but arepresented 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 and filename, located in the lower right corner, directly to the Avaya Solution &Interoperability Test Lab at [email protected].

Documents

Avaya CM SES - MP124 Configuration Guide