Avaya Communication Manager SES & AudioCodes MediaPack 118 FXO

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Avaya Communication Manager SES & AudioCodes MediaPack 118 FXO | Manualzz

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring the AudioCodes MP-118

Analog VoIP Gateway Utilizing Both the FXS/FXO Ports with

Avaya SIP Enablement Services and Avaya Communication

Manager - Issue 1.0

Abstract

These Application Notes describe the procedures for configuring the AudioCodes MP-118

Analog VoIP Gateway with Avaya SIP Enablement Services and Avaya Communication

Manager.

The AudioCodes MP-118 Analog VoIP Gateway (MP-118) serves as a gateway between legacy analog endpoints/trunks at a branch location and a VoIP infrastructure at a main location using the Session Initiation Protocol (SIP). The MP-118 has 4 FXS (analog endpoint) ports and 4 FXO (POTS trunk) ports. Earlier Application Notes focused on the interoperability of the MP-118 with Avaya SIP Enablement Services and Avaya

Communication Manager in a test configuration utilizing only the FXS ports and is documented in [10]. These current Application Notes describe a test configuration that utilizes both the FXS and FXO ports and are a super-set of the earlier Application Notes.

Information in these Application Notes has been obtained through DeveloperConnection compliance testing and additional technical discussions. Testing was conducted via the

DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab.

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1. Introduction

These Application Notes describe the procedures for configuring the AudioCodes MP-118 Analog

VoIP Gateway with Avaya SIP Enablement Services (SES) and Avaya Communication Manager.

The AudioCodes MP-118 Analog VoIP Gateway (MP-118) serves as a gateway between legacy analog endpoints/trunks at a branch location and a VoIP infrastructure at a main location using the

Session Initiation Protocol (SIP). The MP-118 has 4 FXS (analog endpoint) ports and 4 FXO

(POTS trunk) ports. The FXO ports are intended to be used for local inbound and outbound PSTN access for the users located at the branch. Users at the main location will obtain PSTN access from the main location. Thus, a call placed from each location dialed with the same destination digit string will be routed differently based on where the call originates. This requires the use of the

Multiple Locations feature on Avaya Communication Manager. The FXO ports are also used as a failover path if the data WAN is unavailable and SIP calls cannot be made. The data WAN being unavailable and causing a failover can be due to three types of failures at the branch: a failure of the

WAN link, a failure of the LAN providing access to the WAN or power loss of the MP-118. In the case of power loss, the FXS ports are connected directly to the corresponding FXO ports (i.e., FXS port 1 is connected to FXO port 1, FXS port 2 is connected to FXO port 2, etc.).

The FXO ports of the MP-118 will typically connect to POTS trunks from the PSTN which do not provide any far-end disconnect indication. This may cause some undesirable behavior when incoming trunk calls interact with some features like voicemail and conferencing. This is not a limitation of the MP-118 but a limitation of the functionality provided by the trunk. For a more detailed discussion of this topic, see the test results in Section 6.2.

The MP-118 registers with the Avaya SES as a SIP endpoint for each FXS or FXO port which is enabled. On the Avaya SES, the FXS ports are configured as users with media server extensions and the FXO ports are configured as users without media server extensions. Thus, when calls are routed to Avaya Communication Manager, calls from the FXS ports appear as calls from extensions and calls from the FXO ports appear as trunk calls.

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. The main site has an Avaya SES and an Avaya S8300

Server running Avaya Communication Manager in an Avaya G700 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. An

ISDN-PRI trunk connects the media gateway to the PSTN.

The branch site has an AudioCodes MP-118 Analog VoIP Gateway with two analog telephones, and a fax machine. The branch site also has two Avaya 4600 Series IP Telephones (with SIP firmware).

The MP-118 connects the branch site to the PSTN via an FXO (POTS) trunk. The other three FXO ports were not connected. At the branch site, all SIP telephones and analog endpoints (MP-118 FXS ports) are registered directly to Avaya SES and are administered as Outboard Proxy SIP (OPS) stations on Avaya Communication Manager. The PSTN trunk (MP-118 FXO port) is registered with

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the SES but does not have an OPS station administered on Avaya Communication Manager. The

SIP telephones and the MP-118 use the IP network gateway as the default gateway. The MP-118 only supports the operation of the analog endpoints and is not required for the SIP telephones. As a result, if the data WAN is unavailable, the MP-118 allows the analog telephones to keep functioning but the SIP telephones will not.

The PSTN numbers assigned to the ISDN-PRI trunk at the main site are mapped to telephone extensions at the main site. The PSTN number assigned to the POTS line at the branch site is mapped to an extension at the branch site.

Figure 1: MP-118 Test Configuration

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1.2. Operation

As mentioned previously, PSTN calls are routed based on the dialed digits as well as where the call originates. The Multiple Location feature and Automatic Route Selection (ARS) are used to provide this functionality. In addition, two SIP trunk groups are created between Avaya Communication

Manager and Avaya SES. The first trunk group is the initial SIP trunk group required of any SIP installation. It is associated with a SIP signaling group with the far-end domain set to the domain of the Avaya SES. The second trunk group is associated with a signaling group that uses the IP address of the MP-118 as the far-end domain. This second domain information is passed to Avaya SES in the

SIP signaling messages for calls using this second trunk group. This information allows the Avaya

SES to route these calls to the MP-118.

The call flow for an inbound and outbound call from the MP-118 is described below using this configuration. It should be noted that even though the call originates and terminates on the MP-118, the call is sent to Avaya Communication Manager for processing. This allows Avaya

Communication Manager to apply access and routing criteria, as well as make available a wide range of PBX features to the call.

Outbound call to the PSTN using the MP-118 FXO ports:

1. An analog telephone connected to the MP-118 at the branch dials an ARS access code + a

PSTN number.

2. The MP-118 initiates a SIP call to the Avaya SES with the SIP domain specified in the

Avaya SES. This is the same SIP domain with which the endpoint registered (e.g., business.com).

3. The Avaya SES recognizes the call is from a registered user in its own domain which has a media server extension assigned to it. Thus, Avaya SES routes the call automatically to

Avaya Communication Manager.

4. Based on the access code, Avaya Communication Manager uses ARS to route the call.

Based on the location where the call originates, the Avaya Communication Manager will select the proper ARS table unique to that location. This ARS table will point to a route pattern specific for that location.

5. The route pattern will point to the SIP trunk group that is unique for that location which has the far-end domain set to the IP address of the MP-118.

6. Based on this route pattern, Avaya Communication Manager routes the call back to the

Avaya SES using this new trunk group. This has the effect of changing the SIP domain in the SIP signaling messages to the IP address of the MP-118 instead of business.com.

7. The Avaya SES receives the call and determines it is not for its own domain (business.com) so forwards it to the new domain (the MP-118 IP address).

8. The MP-118 receives the call with the PSTN number and routes it to an available trunk.

Inbound call from the PSTN using the MP-118 FXO ports:

1. A PSTN caller dials a PSTN number that is associated with one of the POTS trunks connected to the MP-118. The MP-118 maps this incoming call to an extension which can be located at either site. This example assumes the mapped extension is one of the analog telephones connected to the MP-118. The MP-118 will initiate a SIP call to the Avaya SES for this extension.

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2. The Avaya SES recognizes the call is for a registered user in its own domain which has a media server extension assigned to it. Thus, Avaya SES routes the call automatically to

Avaya Communication Manager.

3. The Avaya Communication Manager receives the call and determines that the destination extension is an OPS station associated with the SIP trunk connected to the Avaya SES. The

Avaya Communication Manager then routes the call back to the Avaya SES.

4. The Avaya SES then routes the call to the registered IP address for that extension (the MP-

118).

2. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration provided:

Equipment Software/Firmware

Avaya S8300 Server with Avaya G700 Media

Gateway

Avaya SIP Enablement Services (SES)

Avaya 4602SW IP Telephone

Avaya 4620SW IP Telephones

Avaya 4625SW IP Telephone

Avaya Communication Manager 3.1.3

(R013x.01.3.640.2)

3.1.1

SIP version 2.2.2

Avaya 6408D Digital Telephone

Analog Telephones

H.323 version 2.7

-

-

Analog Fax Machines

Windows PCs

AudioCodes MP-118 Analog VoIP Gateway

-

Windows XP Professional

5.00A.011.008

3. Configure Avaya Communication Manager

This section describes the Avaya Communication Manager configuration. It is divided into three parts. The first part is the initial SIP configuration required of any SIP installation. The second part describes the configuration needed for multiple location support and call routing based on origination location. The third part includes the configuration of OPS stations.

The following configuration of Avaya Communication Manager was performed using the System

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

translation command to make the changes permanent.

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3.1. Initial SIP Configuration

This section describes the initial configuration required on the Avaya Communication Manager to support SIP.

Step Description

sufficient SIP trunk capacity exists. On Page 2, verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed. Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. Thus, a call from a SIP telephone to another SIP telephone will use two SIP trunks. A call between a non-SIP telephone and a SIP telephone will only use one trunk. In this solution, each analog endpoint at the branch counts as a SIP telephone.

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. display system-parameters customer-options Page 2 of 10

OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 100 10

Maximum Concurrently Registered IP Stations: 20 0

Maximum Administered Remote Office Trunks: 0 0

Maximum Concurrently Registered Remote Office Stations: 0 0

Maximum Concurrently Registered IP eCons: 0 0

Max Concur Registered Unauthenticated H.323 Stations: 0 0

Maximum Video Capable H.323 Stations: 0 0

Maximum Video Capable IP Softphones: 0 0

Maximum Administered SIP Trunks: 100 24

Maximum Number of DS1 Boards with Echo Cancellation: 0 0

Maximum TN2501 VAL Boards: 0 0

Maximum G250/G350/G700 VAL Sources: 5 1

Maximum TN2602 Boards with 80 VoIP Channels: 0 0

Maximum TN2602 Boards with 320 VoIP Channels: 0 0

Maximum Number of Expanded Meet-me Conference Ports: 10 0

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

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Step Description

2. In order to support SIP the following features must be enabled. Use the display

system-parameters customer-options command to verify that the following fields have been set to y.

Page 4: Enhanced EC500? y

Page 4: ISDN-PRI? y

Page 4: IP trunks? y

In addition, the compliance test used Automatic Route Selection (ARS) and the

Multiple Locations feature to route PSTN bound calls (11 digit numbers) between the branch and main sites. Thus, the following features must also be enabled.

Page 3: ARS? y

Page 5: Multiple Locations? y

If a required feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes. the Avaya SES. In this case, SES and 10.75.5.6 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 10.75.5.2 are the name and IP address assigned to the Avaya S8300 Server. change node-names ip Page 1 of 1

IP NODE NAMES

Name IP Address Name IP Address

SES 10 .75 .5 .6 . . . default 0 .0 .0 .0 . . . myaudix 10 .75 .5 .7 . . .

procr 10 .75 .5 .2 . . .

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Step Description

to be changed, to define the connectivity settings for all VoIP resources and IP endpoints within the region. Select an IP network region that will contain the Avaya

SES 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 7. In the case of the compliance test, the same IP network region that contains the Avaya S8300 Server and

Avaya IP Telephones was selected to contain the Avaya SES server. By default, the

Avaya S8300 Server and IP telephones are in IP network region 1.

On the IP Network Region form:

The Location field is set to 1. This associates this IP network region with location 1.

The Authoritative Domain field is configured to match the domain name configured on Avaya SES. In this configuration, the domain name is

business.com. This name will appear in the “From” header of SIP messages originating from this IP region.

Enter a descriptive name for the Name field.

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 the

Avaya G700 Media Gateway. This is true for both intra-region and inter-region

IP-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 calls within this IP network region. If different IP network regions are used for the

Avaya S8300 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 Audio PHB Value is 46, which translate to a DiffServ header value of

0xb8.

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: 1

Authoritative Domain: business.com

Name: default

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: 3327

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

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Step Description

Step 4, to enter the supported audio codecs for calls routed to Avaya SES. Multiple codecs can be listed in priority order to allow the codec 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 voice quality. The example below shows the values used in the compliance test. 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: machines. The Modem field should be set to off. The screen below shows the setting used for the fax testing. 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 3

Clear-channel n 0

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Step Description

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 between

Avaya SES and Avaya Communication Manager.

Specify the Avaya S8300 Server (node name procr) and the Avaya SES Server

(node name SES) 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 3. For alternative configurations that use a C-LAN board, the near (local) end of the SIP signaling group will be the C-LAN board instead of the Avaya S8300 Server.

Ensure that the recommended TLS port value of 5061 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 4. This defines which IP network 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 business.com. This domain is specified in the Uniform Resource Identifier (URI) of the SIP “To” header in the INVITE message.

The Direct IP-IP Audio Connections field is set to y.

The DTMF over IP field must be set to the default value of rtp-payload for a

SIP trunk. This value enables Avaya Communication Manager to send DTMF transmissions using RFC 2833.

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

SIGNALING GROUP

Group Number: 1

Group Type: sip

Transport Method: tls

Near-end Node Name: procr Far-end Node Name: SES

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

Far-end Network Region: 1

Far-end Domain: business.com

Bypass If IP Threshold Exceeded? n

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

IP Audio Hairpinning? n

Session Establishment Timer(min): 120

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Step Description

8. Add a SIP trunk group by using the add trunk-group n command, where n 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 7.

Specify the Number of Members supported by this SIP trunk group. As mentioned earlier, each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. Thus, a call from a SIP telephone to another SIP telephone will use two SIP trunks. A call between a non-SIP telephone and a SIP telephone will only use one 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: SES Trk Grp COR: 1 TN: 1 TAC: 101

Direction: two-way Outgoing Display? n

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Signaling Group: 1

Number of Members: 24

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. 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

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Step Description

10. Use the change public-unknown-numbering 0 command to define the full calling party number to be sent to the far-end. Add an entry for the trunk group defined in

Step 8. In the example shown below, all calls originating from a 5-digit extension beginning with 3 and routed across trunk group 1 will be sent as a 5 digit calling number. This calling party number will be sent to the far-end in the SIP “From” header. change public-unknown-numbering 0 Page 1 of 2

NUMBERING - PUBLIC/UNKNOWN FORMAT

Total Total

Ext Ext Trk CPN CPN Ext Ext Trk CPN CPN

Len Code Grp(s) Prefix Len Len Code Grp(s) Prefix Len

5 3 1 5

5 3 99 5

11. Create a route pattern that will use the SIP trunk that connects to Avaya SES. This route pattern will be used as a default route for SIP calls in Step 12. Some transfer scenarios using alphanumeric handles (i.e., user names) instead of extensions require a default route pattern. These call scenarios were not tested as part of the compliance test, however, the creation of this default route pattern is included here for completeness.

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. change route-pattern 1 Page 1 of 3

Pattern Number: 3 Pattern Name: SIP

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 user

6: n user

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

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Step Description

12. Use the change locations command to assign the default SIP route pattern to the location. This location uses the default name of Main and is shown in the example below. Enter the route pattern number from the previous step in the Proxy Sel. Rte.

Pat. field. The default values may be retained for all other fields. change locations Page 1 of 4

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: :

13. ARS is used to route calls to the PSTN. If the Multiple Location feature is enabled,

ARS supports multiple ARS Digit Analysis Tables. A separate table is supported for each location as well as a general table which is used if a matching entry cannot be found in the location specific table. In the compliance test, the location 1 table was blank so that location 1 (the main site) would use the general table for ARS routing.

PSTN numbers that begin with 1732 were used for testing.

Use the change ars analysis n command to add an entry in the general table for the dialed string beginning with n. The general table is shown below as indicated by the

Location field being set to all. In the example shown, PSTN numbers that begin with

1732 and 11 digits long use route pattern 2. Route pattern 2 routes calls to the ISDN-

PRI trunk between the main site and the PSTN shown in Figure 1. The configuration of the PRI trunk is beyond the scope of these Application Notes. change ars analysis 1732 Page 1 of 2

ARS DIGIT ANALYSIS TABLE

Location: all Percent Full: 3

Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd

1732 11 11 2 fnpa n

174 11 11 deny fnpa n

175 11 11 deny fnpa n

176 11 11 deny fnpa n

177 11 11 deny fnpa n

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Step Description

14. To map a PSTN number to a station at the main or branch office, use the change inc-

call-handling-trmt trunk-group n command, where n is the trunk group number connected to the PSTN from the Avaya G700 Media Gateway. The compliance test used trunk group 2 to connect to the PSTN. This trunk group configuration is not shown in these Application Notes. The example below shows two incoming 11-digit numbers being deleted and replaced with the extension number of the desired station. change inc-call-handling-trmt trunk-group 2 Page 1 of 3

INCOMING CALL HANDLING TREATMENT

Service/ Called Called Del Insert

Feature Len Number

tie 11 17325551240 11 30104

tie 11 17325551241 11 30105

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3.2. Multiple Location Support

This section describes the configuration necessary to define the physical branch location as a second location within Avaya Communication Manager with separate routing tables from the main site.

These steps are not necessary if routing based on the call origination location is not required. In the case where it is not necessary to route calls based on the call origination location, the two physical locations could still be represented by a single logical location within Avaya Communication

Manager.

Step Description

SIP route pattern to the location. In the compliance test, the second location is named

Branch. Enter the same default route pattern number that is used for location 1 named

Main in the Proxy Sel. Rte. Pat. field. The default values may be retained for all other fields. change locations Page 1 of 4

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: Branch + 00:00 0 1

3: : location 2 which represents the branch site. Use the same values shown in Section 3.1,

Step 4 for IP network region 1 with the following exceptions. Set the Location field to

2 and the Name field to a descriptive name for the branch site. change ip-network-region 2 Page 1 of 19

IP NETWORK REGION

Region: 2

Location: 2 Authoritative Domain: business.com

Name: Branch

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: 3327

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

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Step Description

region 2 and IP network region 1 (shown in Section 3.1, Step 6). Locate the entry in the table for src rgn 2 and dst rgn 1 and enter the following values.

codec set: 1

direct WAN: y

Total WAN-BW-limits: :NoLimit

Video WAN-BW-limits: :NoLimit

Use default values for all other fields.

On Page 3, define the codec set to be used when placing calls between IP network region 2 and IP network region 1. Use the same values as shown in Section 3.1, Step

5. change ip-network-region 2 Page 3 of 19

Inter Network Region Connection Management

src dst codec direct Total Video Dyn

rgn rgn set WAN WAN-BW-limits WAN-BW-limits Intervening-regions CAC IGAR

2 1 1 y :NoLimit :NoLimit n

2 2 1

2 3

2 4

4. Repeat the previous step for IP network region 1 shown in Section 3.1, Step 4. Locate the entry in the table for src rgn 1 and dst rgn 2 and enter the same values as shown in

Step 3. assigned to IP network region 2. Any IP address not defined in the table defaults to IP network region 1. For the compliance test, all IP addresses on the 192.168.1.0 network are located at the branch site. Enter 192.168.1.1 in the From IP Address field. Enter

192.168.1.254 in the To IP Address field. Set the Region field to 2. change ip-network-map Page 1 of 32

IP ADDRESS MAPPING

Emergency

Subnet Location

From IP Address (To IP Address or Mask) Region VLAN Extension

192.168.1 .1 192.168.1 .254 2 n

. . . . . . n

. . . . . . n

. . . . . . n

. . . . . . n

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Step Description

6. Create a new SIP signaling group using the same procedure as shown in Section 3.1,

Step 7. Use the same parameters with the following exception. Set the Far-end

Domain field to the IP address of the MP-118. This signaling group is used by PSTN outbound calls that are routed back to the MP-118 for accessing the PSTN via the MP-

118 FXO ports. The compliance test used signaling group 5 as shown below. add signaling-group 5

SIGNALING GROUP

Group Number: 5 Group Type: sip

Transport Method: tls

Near-end Node Name: procr Far-end Node Name: SES

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

Far-end Network Region: 1

Far-end Domain: 192.168.1.60

Bypass If IP Threshold Exceeded? n

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

IP Audio Hairpinning? n

Session Establishment Timer(min): 120

7. Create a new trunk group using the same procedure as shown in Section 3.1, Step 8 -

9. Use the same parameters with the following exceptions. Use unique values for the

Group Name and TAC fields. Set the Signaling Group field to the signaling group number created in the previous step. The compliance test used trunk group 5 with the following values.

Group Name: Branch local trks

TAC: 105

Signaling Group: 5

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Step Description

party number to be sent to the far-end. Add an entry for the trunk group defined in

Step 7. In the example shown below, all calls originating from a 5-digit extension beginning with 3 and routed across trunk group 5 will be sent as a 5-digit calling number. This calling party number will be sent to the far-end in the SIP “From” header. change public-unknown-numbering 0 Page 1 of 2

NUMBERING - PUBLIC/UNKNOWN FORMAT

Total Total

Ext Ext Trk CPN CPN Ext Ext Trk CPN CPN

Len Code Grp(s) Prefix Len Len Code Grp(s) Prefix Len

5 3 1 5

5 3 5 5

5 3 99 5

9. Create a route pattern for use by ARS when routing PSTN calls to the branch site.

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 in Step 7. 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 Prefix Mark (Pfx Mrk) field is set to 1. The Prefix

Mark defines how to process the 1 on user dialed 1 + 10 digit calls. If the Prefix Mark is set to 1, then the user dialed 1 is not suppressed and all 11 digits are sent to the far end. The default values may be retained for all other fields. change route-pattern 5 Page 1 of 3

Pattern Number: 3 Pattern Name: Branch Trks

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: 5 0 1 n user

2: n user

3: n user

4: n user

5: n user

6: n user

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

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Step Description

10. Use the change ars analysis n location 2 command to add an entry in the location 2 specific ARS Digit Analysis Table for the dialed string beginning with n. In the example shown, PSTN numbers that begin with 1732 and 11 digits long use route pattern 5. Set the Call Type field to fnpa. Route pattern 5 routes calls to the second

SIP trunk between the main site and the branch with the far-end domain set to the IP address of the MP-118. change ars analysis 1732 location 2 Page 1 of 2

ARS DIGIT ANALYSIS TABLE

Location: 2 Percent Full: 3

Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd

1732 11 11 5 fnpa n

n

n

n

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3.3. OPS Configuration

This section describes the configuration of OPS stations, which is required for each analog endpoint connected to the MP-118 and each SIP endpoint. These Application Notes assume that all necessary configuration has been performed for the SIP endpoints at the main location including the creation of

OPS stations. This section will only focus on the endpoints at the branch.

Step Description

1. All SIP stations are configured as OPS stations on Avaya Communication Manager.

This includes the analog telephones, and fax machine connected to the FXS ports of the

MP-118, which appear as SIP stations to Avaya Communication Manager.

Use the display system-parameters customer-options command to verify Avaya

Communication Manager has sufficient OPS capacity available to add the OPS stations needed for the SIP and analog endpoints at the branch office in Figure 1. If there is insufficient capacity, contact an authorized Avaya sales representative or business partner to make the appropriate changes. display system-parameters customer-options Page 1 of 10

OPTIONAL FEATURES

G3 Version: V13

Location: 1 RFA System ID (SID): 1

Platform: 13 RFA Module ID (MID): 1

USED

Platform Maximum Ports: 900 121

Maximum Stations: 450 41

Maximum XMOBILE Stations: 0 0

Maximum Off-PBX Telephones - EC500: 50 0

Maximum Off-PBX Telephones - OPS: 50 23

Maximum Off-PBX Telephones - SCCAN: 0 0

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Step Description

2. To add a station, use the add station n command where n is an unused extension number. In the Type field, enter the set type. For the analog endpoints connected to the

MP-118, a set type of 4620 was used. Enter IP in the Port field. Enter a descriptive name in the Name field. The default values may be retained for all other fields. add station 30108 Page 1 of 4

STATION

Extension: 30108 Lock Messages? n BCC: 0

Type: 4620 Security Code: TN: 1

Port: IP Coverage Path 1: 1 COR: 1

Name: Harry Coverage Path 2: COS: 1

Hunt-to Station:

STATION OPTIONS

Loss Group: 19 Personalized Ringing Pattern: 1

Message Lamp Ext: 30108

Speakerphone: 2-way Mute Button Enabled? y

Display Language: english Expansion Module? n

Survivable GK Node Name:

Survivable COR: internal Media Complex Ext:

Survivable Trunk Dest? y IP SoftPhone? n

Customizable Labels? y appearance to be used for either an incoming or outgoing call. In the compliance test, each station had voicemail coverage provided by Avaya IA 770 Intuity Audix as reflected in the example below. However, the configuration of this functionality is beyond the scope of these Application Notes. add station 30108 Page 2 of 4

STATION

FEATURE OPTIONS

LWC Reception: audix Auto Select Any Idle Appearance? n

LWC Activation? y Coverage Msg Retrieval? y

LWC Log External Calls? n Auto Answer: none

CDR Privacy? n Data Restriction? n

Redirect Notification? y Idle Appearance Preference? n

Per Button Ring Control? n Bridged Idle Line Preference? n

Bridged Call Alerting? y Restrict Last Appearance? n

Active Station Ringing: single Conf/Trans on Primary Appearance? n

EMU Login Allowed? n

H.320 Conversion? n Per Station CPN - Send Calling Number?

Service Link Mode: as-needed

Multimedia Mode: enhanced

MWI Served User Type: Display Client Redirection? n

AUDIX Name: IA770 Select Last Used Appearance? n

Coverage After Forwarding? s

Direct IP-IP Audio Connections? y

Emergency Location Ext: 30108 Always Use? n IP Audio Hairpinning? n

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Step Description

supported by the endpoint. To create a call appearance, enter call-appr as the button assignment. The example below shows the configuration of one of the analog endpoints connected to the MP-118. The analog endpoints used in the compliance test were all configured with two call appearances.

There are two Feature Name Extensions (FNEs) that require the assignment of feature buttons in order to operate. The Automatic Callback FNE requires the assignment of an

auto-cback button. The Conference On Answer FNE requires the assignment of a no-

hld-cnf button. Both of these button assignments are shown in the example below. add station 30108 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: List2: List3:

BUTTON ASSIGNMENTS

1: call-appr 5: auto-cback

2: call-appr 6: no-hld-cnf

3: 7:

4: 8:

5. Map the Avaya Communication Manager extension to the Avaya SES media server extension defined in Section 4, Step 6 with the add off-pbx-telephone station-

mapping command. Enter the values as shown below:

Station Extension: Avaya Communication Manager extension

Application: OPS

Phone Number: Avaya SES media server extension

Trunk Selection: The SIP trunk group number defined in Section 3.1.

Configuration Set: Enter a valid configuration set which contain the default values. 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

30108 OPS - 30108 1 1

-

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Step Description

in Step 4. Verify that the Mapping Mode is set to both. This setting allows the OPS station to both originate and terminate calls. add off-pbx-telephone station-mapping Page 2 of 2

STATIONS WITH OFF-PBX TELEPHONE INTEGRATION

Station Call Mapping Calls Bridged

Extension Limit Mode Allowed Calls

30108 2 both all both office has five user endpoints: two analog telephones connected to the MP-118

(x30108 and x30109), two Avaya 4600 Series SIP Telephones (x30101 and x30102), and a fax machine (x30110).

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4. Configure Avaya SES

This section covers the configuration of Avaya SES. Avaya SES is configured via an Internet browser using the administration web interface. It is assumed that the Avaya SES software and the license file have already been installed on the server. During the software installation, an installation script is run from the Linux shell of the server to specify the IP network properties of the server along with other parameters. In addition, it is assumed that the Setup screens of the administration web interface have been used to initially configure the Avaya SES. For additional information on these installation tasks, refer to [5].

Step Description

1. Access the Avaya SES administration web interface by entering http://<ip-addr>/admin as the URL in an Internet 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.

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

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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 on the 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 the following steps.

4. As part of the Avaya SES installation and initial configuration procedures, the following parameters were defined. Although these procedures are out of the scope of these Application Notes, the values used in the compliance test are shown below for reference. After each parameter is a brief description of how to view the value from the Avaya SES Administration Home Page shown in the previous step.

• SIP Domain: business.com

(To view, navigate to Server Configuration System Parameters)

• Host (SES IP address): 10.75.5.6

(To view, navigate to Host List; Click Edit)

• Media Server (Avaya Communication Manager) Interface Name: CMeast

(To view, navigate to Media Server List; Click Edit)

• SIP Trunk IP Address (Avaya S8300 Server IP address): 10.75.5.2

(To view, navigate to Media Server List; Click Edit)

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Step Description

5. A user must be added on Avaya SES for each of the extensions at the branch office created on Avaya Communication Manager in Section 3.3, Steps 2 – 6. From the left pane, navigate to Users Add. Enter the values as shown below.

Primary Handle: Enter the extension for this user.

Password: Enter a valid password for logging into the SIP endpoint.

Confirm Password: Re-enter the password.

Host: Select the Avaya SES server from the pull-down menu.

First Name: Any descriptive name.

Last Name: Any descriptive name.

Check the Add Media Server Extension checkbox. Click the Add button to proceed.

A confirmation window will appear. Click Continue on this new page to proceed.

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Step Description

same extension used in the previous step. In the Media Server field, select from the pull-down menu the name of the media server shown in Step 4.

Click the Add button to complete the operation.

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Step Description

8. In addition, a user must be added on Avaya SES for each of the FXO (trunk) ports on the MP-118. These users will not have media server extensions assigned to them.

From the left pane, navigate to Users Add. Enter the values as shown below.

Primary Handle: Enter the PSTN number for this user.

Password: Enter a valid password for this user.

Confirm Password: Re-enter the password.

Host: Select the Avaya SES server from the pull-down menu.

First Name: Any descriptive name.

Last Name: Any descriptive name.

Do not check the Add Media Server Extension checkbox. Click the Add button to proceed. A confirmation window will appear. Click Continue on this new page to proceed.

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Step Description

10. Calls from the FXO ports on the MP-118 to any SIP endpoint will be routed to Avaya

Communication Manager automatically since the called party is a registered SES user and has a media server extension associated with it. Calls from the FXO ports on the

MP-118 to a non-SIP endpoint requires an Media Server Address Map to route the call to Avaya Communication Manager since neither the caller or called party are registered

SES users with media server extensions. To cover this particular case in the compliance test, a Media Server Address Map was defined that would route any call to extensions 3xxxx to Avaya Communication Manager. Even though the address map is defined quite broadly, this extension range also includes the SIP endpoints, it functions properly because it only gets applied to calls which the Avaya SES does not know how to route automatically.

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 below.

Click on the Map link.

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Step Description

11. On the List Media Server Address Map page, click on the Add Map In New Group link.

12. On the Add Media Server Address Map page that appears:

Enter a descriptive name in the Name field.

In the Pattern field, enter an expression to define the matching criteria for calls to be routed to the main site from the MP-118. The example below shows the expression used in the compliance test. This expression will match an URI that begins with sip:3 followed by any digit between 0-9 for the next 4 digits.

Appendix A contains additional information on the syntax used for address map patterns.

Click the Add button.

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Step Description

13. After configuring the Media Server Address Map, the initial Contact information is populated automatically and directs the calls to the IP address of the Avaya Media

Server (10.75.5.2) 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)@10.75.5.2:5061;transport=tls

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5. Configure the MP-118

This section describes the procedures for configuring the MP-118. These procedures assume the

MP-118 has been installed using the procedures documented in [7] and has been assigned an IP address. In addition, the procedure shows the configuration of all FXO and FXS ports even though not all ports were connected in the compliance test.

Step Description

1. The configuration of the MP-118 is done via a Web browser. To access the device, enter the IP address of the MP-118 in the Address field of the browser.

2. The following pop-up window will appear. Login with the proper credentials.

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Step Description

3. The MP-118 main page will appear as shown below.

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Step Description

4. The network settings that were configured during installation can be viewed by selecting Advanced Configuration in the left pane then navigating to Network

Settings

IP Settings in the right pane. If necessary, changes can be made to the settings on this page followed by clicking Submit. For the compliance test, the IP

Address, Subnet Mask and Default Gateway Address were set to values consistent with the test configuration shown in Figure 1. The Media Premium QoS must match the Audio PHB Value set on Avaya Communication Manager in Section 3.1, Step 4.

Default values may be retained for all other values.

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Step

5. Protocol Definition

Description

To access these parameters, select Protocol Management in the left pane and navigate to Protocol Definition in the right pane. The pull-down choices for Protocol

Definition are shown below.

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Step

6. Proxy and Registration

Description

From the menu shown in Step 5, navigate to Protocol Definition

Proxy &

Registration. Configure the parameters as described below. If DNS is used, fully qualified domain names (FQDN) can be used in fields requesting IP addresses below.

For the Enable Proxy field, select Use Proxy from the pull-down menu.

In the Proxy IP Address field, enter the IP address of the Avaya SES.

For the Enable Proxy Keep Alive field, select Using Options from the pulldown menu. The MP-118 will use the SIP OPTIONS message as a handshake mechanism with the Avaya SES to determine if the SIP connection is up. If the connection is down, the MP-118 will failover to the FXO ports.

Set the Enable Fallback to Routing Table field to Enable.

For the Always Use Proxy field, select Enable.

For the Send All Invite to Proxy field, select Yes. This directs the MP-118 to send all INVITE requests to the Avaya SES, including those generated as the result of a transfer or redirect.

For the Enable Registration field, select Enable. This will allow the MP-118 to register the FXS and FXO ports with the Avaya SES.

In the Registrar IP Address field, enter the IP address of the Avaya SES.

Default values may be retained for all other fields. Scroll down to continue configuring parameters on the lower half of the screen.

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Step Description

7. Configure the parameters as described below.

Set the Registration Time field to a value (in seconds) larger than the minimum required by the Avaya SES. The default value of 180 is too small.

The compliance test used 3600.

In the Gateway Name field, enter the IP address of the Avaya SES.

For the Authentication Mode, select Per Endpoint. The MP-118 will authenticate each FXS/FXO port separately.

Click Submit.

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Step

8. General Protocol Parameters

Description

From the menu shown in Step 5, navigate to Protocol Definition

General

Parameters. Configure the parameters as described below.

For the Enable Early Media field, select Enable. If enabled, the MP-118 sends Session Description Protocol (SDP) information in the 18x responses allowing the media stream to be set-up prior to answering the call.

If fax support is needed, select T.38 Relay as the Fax Signaling Method.

Select No for the Use “user=phone” in SIP URL field.

Default values may be retained for all other fields. Scroll down to the bottom of the page and click Submit (not shown).

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Step

9. Coders

Description

From the menu shown in Step 5, navigate to Protocol Definition Coders. In the screen below, select the list of preferred codecs to be used by the MP-118 with the most preferred codec at the top and working downward to the least preferred. This list must have an overlap with the list provided on Avaya Communication Manager in

Section 3.1, Step 5. The codec is selected from the pull-down menu under the Coder

Name field.

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

MP-118 uses the equivalent of a G.729AB codec. Default values were retained for all other fields.

Click Submit.

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Step

10. DTMF and Dialing

Description

From the menu shown in Step 5, navigate to Protocol Definition

DTMF &

Dialing. Configure the parameters as described below.

In the Max Digits in Phone Num field, enter the maximum number of digits that can be dialed.

For the Declare RFC 2833 in SDP field, select Yes.

For the 1

st

Tx DTMF Option field, select RFC 2833. This selects RFC 2833 as the preferred DTMF transmission method.

Select 127 as the RFC 2833 Payload Type to match the value used by the

Avaya SIP Telephones. Media may not be redirected (shuffled) in all scenarios from Avaya Communication Manager to the endpoints if this value is not the same as the SIP Telephones.

Assign the digit map pattern to the Digit Mapping Rules field. If the dialed digits matches any pattern in the digit map, the MP-118 stops collecting digits and places the call. The digit map may contain up to 52 patterns each separated by a vertical bar (|). The maximum length of the entire digit map is limited to

152 characters. For the compliance test, two patterns were defined. One pattern matches any 5 digit number beginning with 3 (3xxxx). The other pattern matches any 12 digit number starting with 91 (91xxxxxxxxxx)

Default values may be retained for all other fields. Click Submit.

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Step

11. Advanced Parameters

Description

To access these parameters, select Protocol Management in the left pane and navigate to Advanced Parameters in the right pane. The pull-down choices for Advanced

Parameters are shown below.

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Step

12. General Advanced Parameters

Description

From the menu shown in Step 11, navigate to Advanced Parameters

General

Parameters. Configure the parameters as described below.

Select Enable for the Enable Polarity Reversal and Enable Current

Disconnect fields. This will allow the MP-118 to provide the proper disconnect indication to various line types if supported by the far-end.

The Disconnect Call on Silence Detection field may be set to Yes to allow the

MP-118 to disconnect a FXO port if silence is detected for a given period of time. Typically, the POTS trunks connected to the MP-118 FXO ports do not provide a disconnect indication when the far-end hangs up. Depending on the needs of the customer, the Disconnect Call on Silence feature may be required to prevent the FXO ports from being tied up indefinitely when interacting with certain PBX features. See Section 6.2 for more details on this topic. For the purposes of the compliance test, this feature was disabled by setting the

Disconnect Call on Silence Detection field to No. If this field is set to Yes, then the Silence Detection Period and the Silence Detection Method fields must also be configured. Care should be taken when selecting the silence detection period so that a call between two users with minimal conversation is not disconnected by mistake.

Default values may be retained for all other fields. Scroll down to continue configuring parameters on the lower half of the screen.

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Step Description

13. Configure the parameters as described below.

In the Max Number of Active Calls field, a value can be entered to define the number of active calls allowed on the gateway up to the maximum supported by the MP-118. The maximum number supported is equal to the maximum number of ports (FXS + FXO) available on the gateway. The default value of 0 will configure the MP-118 to use the maximum. However, for the compliance test, this field was specifically set to the maximum of 8.

Default values may be retained for all other fields. Click Submit.

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Step

14. Supplementary Services

Description

From the menu shown in Step 11, navigate to Advanced Parameters

Supplementary Services. Configure the parameters as described below.

If the analog phones connected to the MP-118 support Caller ID then set the

Enable Caller ID field to Enable. Caller ID can also be controlled on a per port basis by navigating to Protocol Management

Endpoint Settings

Generate Caller ID to Tel Table (see Step 25).

Select Enable for the Enable MWI and MWI Display fields if the analog phones support a visual MWI indicator. For the compliance test, even though these fields were enabled, MWI was only tested for stutter dial tone.

Hold, Transfer, Call Forwarding and Call Waiting are enabled by default.

Default values may be retained for all other fields. Scroll down to the bottom of the page and click Submit (not shown).

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Step

15. Manipulation Tables

Description

To access these parameters, select Protocol Management in the left pane and navigate to Manipulation Tables in the right pane. The pull-down choices for the

Manipulation Tables are shown below.

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Step Description

16. IP to Telephone Destination Numbers

From the menu shown in Step 15, navigate to Manipulation Tables

IP to Tel

Destination Numbers.

This table defines the digit manipulation that will be performed on calls directed from the IP side of the MP-118 to the analog side. The Destination Prefix, Source Prefix and Source IP columns define which calls to manipulate. The 3 remaining columns define the digit manipulation to apply to these calls. The digit manipulation is applied in a specific order beginning with the Number of Stripped Digits, followed by the

Number of Digits to Leave, ending with the Prefix (Suffix) to Add. The table is ordered from top to bottom. The MP-118 will select the first entry in the table with the correct matching criteria. In the example below, all calls dialed with twelve digits beginning with 91 coming from any source extension and source IP will have the first digit stripped off.

This manipulation is required to allow the branch user to dial the same outbound number when the data WAN is available or not. Under normal conditions, a user dials

9 + the 11-digit PSTN number for an outbound call. The call is forwarded to the

Avaya Communication Manager where the 9 is stripped off before being routed to the

PSTN. If the data WAN is unavailable, the call is still originated as an IP call but cannot be sent to the Main site. The MP-118 will use the table shown below to strip the preceding 9 off of this IP call before routing it to the FXO port for the PSTN.

Click Submit.

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Step

17. Routing Tables

Description

To access these parameters, select Protocol Management in the left pane and navigate to Routing Tables in the right pane. The pull-down choices for the Routing Tables are shown below.

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Step

18. Telephony to IP Routing

Description

From the menu shown in Step 17, navigate to Routing Tables Tel to IP Routing.

This table defines the mapping of analog calls to an IP address that will process the IP leg of the call if the proxy defined in Step 6 (Avaya SES) is not available. The Dest.

Phone Prefix and Source Phone Prefix columns define which calls are mapped to the

IP address in the Dest. IP Address column. If DNS is used, a fully qualified domain name (FQDN) can be used in the Dest. IP Address column.

In the example below, the table entry maps calls with any destination prefix, or any source prefix to IP address 192.168.1.60 which is the address of the MP-118. When the MP-118 processes the call, the MP-118 will use the table in Step 19 to route the call to a FXO port. Thus, Steps 18 – 19 provide the routing necessary to support the failover functionality to the PSTN. The default value may be used for the Profile ID column.

Click Submit.

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Step

19. IP to Hunt Group Routing Table

Description

From the menu shown in Step 17, navigate to Routing Tables

IP to Trunk Group

Routing.

This table defines the mapping of IP calls to a group of channels or “hunt group”. This is necessary when hunt groups are defined in the Endpoint Phone Number Table in

Step 20. In Step 20, the FXS ports are assigned to hunt group 1 and the FXO ports are assigned to hunt group 2. The Dest. Phone Prefix, Source Phone Prefix and Source

IP Address columns define which calls are mapped to the hunt group in the Hunt

Group ID column. In the first entry in the example below, all calls to any extension

30108 - 30111 from any source extension and source IP will be routed to hunt group 1.

The # at the end of the Dest. Phone Prefix field indicates “end of number”. Thus, the dialed number must be an exact match with an extension and not simply a match on the prefix of 30108 - 30111.

The second entry routes any other call to hunt group 2. This entry is needed to route outbound PSTN calls to the FXO ports and to route calls in the failover case when the data WAN is unavailable. If a user dials an outbound number and it cannot reach the main site, this second entry will direct the call to hunt group 2 which contains the FXO ports.

Click Submit.

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Step

20. Endpoint Phone Numbers

Description

Select Protocol Management in the left pane and navigate to Endpoint Phone

Numbers in the right pane.

The Endpoint Phone Number Table maps a particular channel/port to a phone number and hunt group. In the Channel(s) column, enter a range of channels to be assigned.

In the Phone Number column, enter the starting extension for the range of extensions.

In the Hunt Group ID column, enter the hunt group that contains these extensions.

In the example below, the first entry assigns channels 1 – 4 with extensions 30108 –

30111 to hunt group 1. The second entry assigns channel 5 - 8 (the FXO ports) to hunt group 2. The Phone Number field is the PSTN number assigned to the FXO port. In this example, the FXO ports are assigned sequential numbers starting at 7325551234.

Click Submit.

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Step

21. Hunt Group Settings

Description

Select Protocol Management in the left pane and navigate to Hunt Group Settings in the right pane.

Configure the parameters are described below. This is necessary when hunt groups are defined in the Endpoint Phone Number Table in Step 20.

For Hunt Group ID 1 which contain the FXS (endpoint) ports, select the

Channel Select Mode as By Dest Phone Number. Thus, each port in this hunt group will only be selected if its destination phone number is dialed. Select the

Registration Mode to be Per Endpoint.

For Hunt Group ID 2 which contain the FXO (trunk) ports, select the Channel

Select Mode as Cyclic Ascending. The ports in this hunt group are treated as a pool, and each will be selected in cyclic ascending order. Select the

Registration Mode to be Per Endpoint. This allows the MP-118 to register each of the FXO ports.

Click Submit.

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Step

22. Endpoint Settings

Description

To access these parameters, select Protocol Management in the left pane and navigate to Endpoint Settings in the right pane. The pull-down choices for Endpoint Settings are shown below.

23. Authentication

From the menu shown in Step 22, navigate to Endpoint Settings Authentication.

The Authentication page defines a username and password combination for authentication of each MP-118 port with the Avaya SES. Enter a User Name and

Password that matches the values configured on Avaya SES in Section 4, Step 5.

Click Submit.

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Step

24. Automatic Dialing

Description

From the menu shown in Step 22, navigate to Endpoint Settings

Automatic

Dialing.

The Automatic Dialing page provides the mapping of incoming calls on the FXO ports to a branch extension when the data WAN is unavailable. In the example below, each

FXO port is mapped to a different extension at the branch location. The destination extension is placed in the Destination Phone Number column.

Click Submit.

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Step

25. Caller ID

Description

From the menu shown in Step 22, navigate to Endpoint Settings Caller ID. Enter a descriptive name in the Caller ID/Name column.

Click Submit.

26. Enable Caller ID

From the menu shown in Step 22, navigate to Endpoint Settings

Caller ID

Permissions. Caller ID can be enabled on a per port basis. Select Enable in the

Caller ID column for each port.

Click Submit.

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Step

27. FXO Settings

Description

To access these parameters, select Protocol Management in the left pane and navigate to FXO Settings in the right pane. The pull-down choices for FXO Settings are shown below.

28. FXO Settings

From the menu shown in Step 27, navigate to FXO Settings FXO Settings.

Configure the parameters as described below.

Select One Stage for the Dialing Mode. The MP-118 will not prompt the user with a second dial tone and will send all digits in a single request including any access code prefix.

Select Yes for Answer Supervision.

Default values may be retained for all other fields. Click Submit.

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Step

29. Media Settings

Description

To access these parameters, select Advanced Configuration in the left pane and navigate to Media Settings in the right pane. The pull-down choices for Media

Settings are shown below.

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Step

30. Fax/Modem/CID Settings

Description

From the menu shown in Step 29, navigate to Media Settings

Fax/Modem/CID

Settings. Configure the parameters as described below.

Select T.38 Relay for the Fax Transport Mode to support faxing across the data WAN.

Select Enable Bypass for the V.22 – V.34 Modem Transport Types to support modem calls over SIP planned in future Avaya Communication

Manager releases.

Select G711Mulaw for the Fax/Modem Bypass Coder Type.

Default values may be retained for all other fields. Click Submit.

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Step Description

31. For proper modem interoperability over SIP planned for future Avaya Communication

Manager releases, an additional parameter must be changed in the ini file. From a web browser, enter <ip_address>/AdminPage in the Address field where <ip_address> is the IP address of the MP-118. The main page will appear as shown below.

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Step Description

32. 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 MP-118 uses a proprietary value for the payload type of modem calls so they can be distinguished from other G.711mu calls.

33. To save the configuration, return to the administrative web page and select

Maintenance in the left pane. Click the BURN button.

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6. Interoperability Compliance Testing

This section describes the compliance testing used to verify the interoperability of the AudioCodes

MP-118 using both the FXS and FXO ports with Avaya SIP Enablement Services (SES) and Avaya

Communication Manager. This section covers the general test approach and the test results.

6.1. General Test Approach

The general test approach was to make calls through the MP-118 at the branch site using various codec settings and exercising common PBX features. The calls were made to/from the main site, the

PSTN and within the branch site. The same test cases, where applicable, were repeated with a simulated data WAN outage.

6.2. Test Results

The AudioCodes MP-118 successfully passed compliance testing. The following features and functionality were verified when the data WAN was available.

Calls to/from the MP-118 FXS ports and the main site

Calls to/from MP-118 FXS ports and the PSTN (MP-118 FXO ports)

Calls to/from SIP endpoints at the branch and the PSTN (MP-118 FXO ports)

Intra-branch calls with both analog and SIP endpoints

G.711mu and G.729AB codec support

Proper recognition of DTMF transmissions

Local MP-118 support for Hold, Transfer, and Call Waiting

Proper operation of voicemail with message waiting indicators (MWI). For the analog phones, MWI was provided via stutter dial tone.

Call Forwarding provided by Avaya Communication Manager.

Conferencing (Avaya SIP telephone initiates a conference that includes an MP-118 analog endpoint)

Extended telephony features using Avaya Communication Manager Feature Name

Extensions such as Conference On Answer, Call Park, Call Pickup, Automatic Redial and

Send All Calls. For more information on FNEs, please refer to [6].

T.38 fax support

Proper system recovery after a MP-118 restart

The following features and functionality were verified using an MP-118 analog endpoint when a simulated data WAN failure was introduced.

Automatic failover to the POTS line to complete calls to the main site and PSTN using full

11-digit dialing. Incoming calls to the branch are limited to the single POTS number assigned to the branch.

Intra-branch calls with analog endpoints

Local MP-118 support for Hold, Transfer, and Call Waiting

Fax calls to/from PSTN (not T.38)

The following observations were made during the compliance test:

The FXO ports of the MP-118 will typically connect to POTS trunks from the PSTN which do not provide any far-end disconnect indication. This may cause some undesirable behavior when incoming trunk calls interact with some features like voicemail and conferencing. This is not a limitation of the MP-118 but a limitation of the functionality provided by the trunk.

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In a typical person to person call, this is not an issue. The person at each end of the call knows when the call is over and disconnects themselves by going on-hook and thus no farend disconnect indication is required. In the case, of an incoming trunk call that is routed to voicemail, the voicemail system does not know when the caller has completed recording a message and has hung up. The voicemail system continues to record after the caller has hung up. A similar problem occurs if an incoming trunk call is part of a conference call and hangs up. The near-end does not know that the far-end has hung up so the PSTN line is still included in the conference. After some time, the conference call will begin hearing system announcements from the PSTN stating the call has been disconnected and to please hang up.

To compensate for these interactions, the MP-118 can be configured to disconnect based on a set period of silence as discussed in Section 5, Step 12. In this way, the PSTN line will not be tied up indefinitely in these situations but will be released after the designated period of silence. However, care should be taken when selecting the silence detection period so that a call between two users with minimal conversation is not disconnected by mistake. The silence detection period does not affect calls on-hold.

The MP-118 does not support initiating a conference using flash hook. In addition, the

Avaya Communication Manager Conference On Answer feature did not work for this combined Avaya/AudioCodes solution.

When the data WAN is unavailable and an MP-118 analog endpoint calls the PSTN and does an unattended transfer to another MP-118 analog endpoint, the originator of the transfer does not hear ringback.

When using the Call Park feature from an MP-118 analog endpoint via the Avaya

Communication Manager FNE, the user is required to wait a couple of seconds before hanging up after hearing the feature confirmation tones. Otherwise, the MP-118 does not disconnect the call as expected but instead attempts a transfer.

The MP-118 does not support a drop function. Thus, if an analog phone has two calls up

(one active and one on hold), the user can not disconnect only one of them. Going on-hook disconnects both.

An analog phone connected to the MP-118 receives a call from the main site, places it on hold and receives a second call from the main site which is answered. If the far-end of the active call hangs up, both calls are disconnected including the call on-hold.

An analog phone connected to the MP-118 has a single call active and places it on hold. If a second call is not placed on the second call appearance, the initial call is dropped after the dial tone and fast busy tone timeout (approximately 1.5 minutes).

Inbound PSTN callers connected to the FXO ports of the MP-118 do not hear busy tone when calling a “busy” MP-118 FXS endpoint. The caller continues to hear ringback.

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.

• From the Avaya SES web administration interface, verify that all endpoints behind the MP-

118 are registered with the Avaya SES.

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• Verify that calls can be placed to/from the analog endpoints behind the MP-118 and the main branch.

• Verify that calls can be placed to/from the analog endpoints behind the MP-118 and the

PSTN.

• Verify that calls can be placed from the analog endpoints behind the MP-118 when a simulated data WAN failure is introduced.

8. Support

For technical support on the MP-118, contact AudioCodes via the support link at www.audiocodes.com

.

9. Conclusion

These Application Notes describe the procedures required to configure both the FXS and FXO ports of the AudioCodes MP-118 Analog VoIP Gateway to interoperate with Avaya SIP Enablement

Services and Avaya Communication Manager. The FXO ports are used for local PSTN access for the branch location. Earlier Application Notes focused on a configuration which only used the FXS ports of the MP-118 and is documented in [10]. These current Application Notes are a super-set of the earlier Application Notes. The AudioCodes MP-118 successfully passed compliance testing with the observations documented in Section 6.2.

10. Additional References

[1] Feature Description and Implementation For Avaya Communication Manager, Doc # 555-245-

205, Issue 4.0, February 2006.

[2] Administrator Guide for Avaya Communication Manager, Doc # 03-300509, Issue 2.1, May

2006

[3] Avaya Communication Manager Advanced Administration Quick Reference, Doc # 03-300364,

Issue 2, June 2005 Release 3.0

[4] Avaya IA 770 INTUITY AUDIX Messaging Application, Doc # 11-300532, May 2005

[5] Installing and Administering SIP Enablement Services R3.1, Doc# 03-600768, Issue 1.5,

February 2006

[6] Avaya Extension to Cellular and Off-PBX Station (OPS) Installation and Administration Guide

Release 3.0, version 6.0, Doc # 210-100-500, Issue 9, June 2005

[7] LTRT-59802 MediaPack Fast Track Installation Guide, Version 5.0.

[8] LTRT-65406 MediaPack SIP User’s Manual, Version 5.0.

[9] LTRT-65607 MediaPack & Mediant 1000 SIP Analog Gateways Release Notes, Version 5.0.

[10]

Application Notes for Configuring the AudioCodes MP-118 Analog VoIP Gateway with

Avaya SIP Enablement Services and Avaya Communication Manager, Version 1.0, October

2006.

Product documentation for Avaya products may be found at http://support.avaya.com

.

Product documentation for AudioCodes MP-118 Analog VoIP Gateway products may be found at http://www.audiocodes.com

.

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APPENDIX A: Specifying Pattern Strings in Address Maps

The 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, and special

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

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©

2007 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

DeveloperConnection Program at [email protected].

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