Application Notes for AudioCodes MP

Application Notes for AudioCodes MP
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 interoperate with Avaya SIP Enablement Services and
Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 Analog VoIP
Gateway provides voice technology to connect analog telephones, fax machines and modems
to 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 MP124 Analog VoIP Gateway at the customer enterprise sites to register with Avaya SIP
Enablement Services at the service provider data center, effectively making the analog
endpoints appear as SIP endpoints. These analog endpoints were then able to utilize the
telephony features of Avaya Communication Manager.
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.
CTM; Reviewed:
SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
1. Introduction
These Application Notes describe the configuration procedures required for the AudioCodes
MP-124 Analog VoIP Gateway to interoperate with Avaya SIP Enablement Services (SES) and
Avaya Communication Manager via a SIP interface. The AudioCodes MP-124 Analog VoIP
Gateway provides voice technology to connect analog telephones, fax machines and modems to
SIP servers 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
Analog VoIP Gateway at the customer enterprise sites to register with Avaya SIP Enablement
Services server at the service provider data center, effectively making the analog endpoints
appear as SIP endpoints. These analog endpoints were then able to utilize the telephony features
of Avaya Communication Manager.
The AudioCodes MP-124 Analog VoIP Gateway (MP-124) incorporates 24 analog ports for
connection to telephones, fax machines and modems, supporting 24 simultaneous VoIP calls.
1.1. Reference Test Network
Figure 1 shows the reference configuration used for simulating a hosted telephony solution with
two customer enterprise sites and a service provider data center.
The service provider data center site consists of an Avaya SES server, an Avaya G650 Media
Gateway, and an Avaya S8500 Media Server running Avaya Communication Manager and
Avaya IA770 Intuity Audix. The access to the PSTN for all enterprise customers was provided
by an ISDN PRI trunk from the Avaya G650 Media Gateway.
Each enterprise site consists of a MP-124 with the analog ports connected to analog telephones,
and a fax machine or a modem. The MP-124 is connected to the service provider data center
network over an IP network via an Avaya C363T-PWR Converged Stackable Switch. Avaya
4600 Series IP Telephones (H.323) were also located at the enterprise sites to verify calls to and
from the analog endpoints behind the MP-124. These Avaya 4600 Series IP Telephones
registered directly with Avaya Communication Manager.
CTM; Reviewed:
SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Figure 1: Reference Test Configuration
CTM; Reviewed:
SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
2. Equipment and Software Validated
The following equipment and software/firmware were used for the test configuration provided.
Equipment
Avaya S8500 Media Server including Avaya
IA770 Intuity Audix
Avaya G650 Media Gateway
Avaya SIP Enablement Services
Avaya 4600 Series IP Telephones (H.323)
Avaya 363T-PWR Converged Stackable
Switch
AudioCodes MP-124 Analog VoIP Gateway
Analog telephones
Fax machine
PCs with modem
Software/Firmware
Avaya Communication Manager 3.1.2
(R013x.01.2.632.1)
To support T.38 fax, the following
post–GA release was used:
(R013x.01.2.635.0)
25.23.0
3.1
(3.1.0.0-018.0)
2.3 (4620SW)
4.5.14
4.80A.018.007
Windows 2000 Professional
3. Configure Avaya Communication Manager
This section describes the necessary configuration on Avaya Communication Manager to
interoperate with the SIP interface of 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 MP-124 will direct all SIP traffic bound for the
service provider data center to the Avaya SES located there. Avaya SES 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 and
then to the MP-124. As a result, the SIP trunk group configured in this section carries all the SIP
signaling bound to the service provider data center from the MP-124 at the customer enterprise
sites and vice versa.
The following steps were performed at the service provider data center in the configuration
shown in Figure 1.
CTM; Reviewed:
SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
1.
Description
Use the display system-parameters customer-options command to verify that
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 between two analog telephones behind the
MP-124 will use two SIP trunks. A call between an analog telephone behind the
MP-124 and a non-SIP service provider (e.g., PSTN trunk) will only use one SIP trunk.
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
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable H.323 Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum G250/G350/G700 VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
800
2400
0
0
0
0
0
0
800
USED
0
4
0
0
0
0
0
0
200
0
1
0
0
0
0
0
0
0
0
0
0
2 of
10
(NOTE: You must logoff & login to effect the permission changes.)
2.
Use the change node-names 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 other
configuration forms of Avaya Communication Manager. In this example, CLAN and
192.168.10.43 are the name and IP address assigned to the CLAN IP interface on the
Avaya G650 Media Gateway.
change node-names ip
Name
CLAN
MEDPRO
default
procr
ses
CTM; Reviewed:
SPOC 4/19/2007
Page
IP NODE NAMES
IP Address
Name
192.168.10 .43
192.168.10 .42
0 .0 .0 .0
192.168.10 .40
192.168.10 .50
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
1 of
1
IP Address
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
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Step
3.
Description
Use the change ip-network-region n command, where n 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
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 5. In the case of
the 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 Media
Server is in IP network region 1.
On the IP Network Region form:
ƒ 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 from
this IP region.
ƒ 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 G650 Media Gateway. This is true for both intra-region and inter-region
calls. Shuffling can be further restricted at the trunk level on the Signaling
Group form.
ƒ The Codec Set is set to the number of the IP codec set to be used for calls
within this IP network region. In this configuration, this codec set will apply to
all calls to and from the MP-124 as well as any IP telephone within the
enterprise. If different IP network regions are used for the Avaya S8500 Media
Server and the Avaya SES server, then Page 3 of each IP Network Region
form (not shown) must be used to specify the codec set for inter-region
communications.
ƒ The default values can be used for all other fields.
change ip-network-region 1
Page
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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
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
4.
Description
Use the change ip-codec-set n command, where n is the codec set value specified in
Step 3, to enter the supported audio codecs for calls routed to Avaya SES and to the
MP-124. 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. It
must include at least one codec configured on the 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
Codec
1: G.711MU
2: G.729AB
3:
4:
Silence
Suppression
n
n
Frames
Per Pkt
2
2
Packet
Size(ms)
20
20
If T.38 fax is desired, on Page 2 enter t.38-standard in the FAX Mode field. For data
modem calls, retain the default value of off for the Modem Mode field.
change ip-codec-set 1
Page
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2
IP Codec Set
Allow Direct-IP Multimedia? n
FAX
Modem
TDD/TTY
Clear-channel
CTM; Reviewed:
SPOC 4/19/2007
Mode
t.38-standard
off
US
n
Redundancy
0
0
3
0
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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Step
5.
Description
Use the add signaling group n command, where n is the number of an unused
signaling group, to create the SIP signaling group as follows:
ƒ Set the Group Type field to sip.
ƒ 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.
ƒ Specify the node name CLAN on Avaya S8500 Media Server and the node
name 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 configurations that do not use a CLAN board, the near (local) end of the SIP
signaling group will be the procr instead of the CLAN board.
ƒ Ensure that the TLS port value of 5061 is configured in the Near-end Listen
Port and the Far-end Listen Port fields.
ƒ 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 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.
ƒ 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 “To”
header in the INVITE message.
ƒ The Direct IP-IP Audio Connections field is set to y to enable the direct audio
path (shuffling) between the two IP endpoints.
ƒ 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.
ƒ The default values for the other fields may be used.
add signaling-group 1
Page
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1
SIGNALING GROUP
Group Number: 1
Group Type: sip
Transport Method: tls
Near-end Node Name: CLAN
Near-end Listen Port: 5061
Far-end Node Name: ses
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
CTM; Reviewed:
SPOC 4/19/2007
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? y
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
6.
Description
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:
ƒ Set the Group Type field to sip.
ƒ Choose a descriptive Group Name.
ƒ Specify an available trunk access code (TAC) that is consistent with the
existing dial plan.
ƒ Set the Service Type field to tie.
ƒ Specify the signaling group associated with this trunk group in the Signaling
Group field as previously specified in Step 5.
ƒ Specify the Number of Members supported by this SIP trunk group. As
mentioned 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 MP-124, will use
two SIP trunks. A call between a SIP extension and the PSTN will only use
one SIP trunk.
ƒ The default values may be retained for the other fields.
add trunk-group 1
Page
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21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
1
siproute
two-way
n
0
tie
Group Type: sip
COR: 0
Outgoing Display? n
TN: 1
CDR Reports: y
TAC: 701
Night Service:
Auth Code? n
Signaling Group: 1
Number of Members: 100
7.
On Page 3:
ƒ Set the Numbering Format field to public. This field specifies the format of
the calling party number sent to the far-end.
ƒ The default values may be retained for the other fields.
add trunk-group 1
TRUNK FEATURES
ACA Assignment? n
Page
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21
Measured: none
Maintenance Tests? y
Numbering Format: public
Prepend '+' to Calling Number? n
Replace Unavailable Numbers? n
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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Step
8.
Description
Use the change route-pattern command to configure a route pattern that routes to the
outbound SIP proxy server (Avaya SES server). This will be assigned on the locations
screen. Refer to Step 9.
The example below shows 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 configured in Step 6. The FRL field defines the facility restriction level for
this route pattern. The value of 0 is the least restrictive. Default values for all other
fields can be used.
change route-pattern 1
Pattern Number: 1
Pattern Name: siproute
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 1
0
2:
3:
4:
5:
6:
9.
Page
1 of
3
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
Use the change locations command to assign the route pattern to the location. This
screen allows for each location to point to the route pattern that routes to the 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 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
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16
LOCATIONS
ARS Prefix 1 Required For 10-Digit NANP Calls? y
Loc. Name
No.
1:
Main
2:
3:
4:
CTM; Reviewed:
SPOC 4/19/2007
Timezone Rule
Offset
+ 00:00
0
:
:
:
NPA
ARS
FAC
Attd
FAC
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
Prefix
Proxy Sel.
Rte. Pat.
1
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3.2. Configure SIP Stations
The analog endpoints (e.g., analog telephones, fax machines and modems) behind the MP-124
are considered SIP stations from the point of view of Avaya Communication Manager. This
section describes the steps for configuring SIP stations.
SIP stations are administered as off-PBX station (OPS) stations. Many advanced PBX features
are available to OPS stations such as call park, call pickup, and priority calls. These features are
activated by dialing feature access codes (FAC) or by dialing special extensions known as
feature name extensions (FNE). For more details, see [6] and [7]. The configuration for feature
access codes and feature name extensions is not in the scope of this document.
Step
1.
Description
Use the display system-parameters customer-options command to verify that
Maximum Off-PBX Telephones – OPS has been set to a value that will
accommodate the number of telephones to be supported.
display system-parameters customer-options
OPTIONAL FEATURES
G3 Version: V13
Location: 1
Platform: 12
1 of
10
RFA System ID (SID): 1
RFA Module ID (MID): 1
Platform Maximum Ports:
Maximum Stations:
Maximum XMOBILE Stations:
Maximum Off-PBX Telephones - EC500:
Maximum Off-PBX Telephones OPS:
Maximum Off-PBX Telephones - SCCAN:
CTM; Reviewed:
SPOC 4/19/2007
Page
3200
2400
0
2400
2400
0
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
USED
231
21
0
0
16
0
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Step
2.
Description
Use the add station x command, where x is an unused extension, to create a station
extension as follows:
ƒ Leave the default value the for Type field to 6408D+.
ƒ Set the Port field to X. The value X implies that this station is being
administered without hardware, such as a port on an Avaya analog or a digital
circuit pack.
ƒ Set the Name field to any alpha-numeric name.
The default values may be retained for all other fields.
add station 2346241
Page
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4
STATION
Extension:
Type:
Port:
Name:
2346241
6408D+
X
CM CUST A Analog 2
STATION OPTIONS
Loss Group:
Data Module?
Speakerphone:
Display Language:
2
n
2-way
english
Lock Messages? n
Security Code:
Coverage Path 1:
Coverage Path 2:
Hunt-to Station:
BCC:
TN:
COR:
COS:
0
1
1
1
Personalized Ringing Pattern: 1
Message Lamp Ext: 2346241
Mute Button Enabled? y
Media Complex Ext:
IP SoftPhone? n
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
3.
Description
On Page 3, assign the buttons under BUTTON ASSIGNMENTS as follows:
ƒ Make sure that at least two call-appr buttons are administered.
ƒ Assign a no-hld-cnf button to enable the station to activate the Conference on
Answer feature.
ƒ Assign an auto-cback button to enable the station to activate the Automatic
CallBack feature.
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:
Jack:
Cable:
Floor:
Building:
Headset?
Speaker?
Mounting:
Cord Length:
Set Color:
ABBREVIATED DIALING
List1: group
1
BUTTON ASSIGNMENTS
1: call-appr
2: call-appr
3:
4: no-hld-cnf
4.
List2:
n
n
d
0
List3:
5: auto-cback
6:
7:
8:
Use the add off-pbx-telephone station-mapping command to associate an
administered extension with an external phone number as follows:
ƒ Set the Station Extension field to the extension administered in Step 2.
ƒ Set the Application field to OPS.
ƒ In the Phone Number field, enter the phone number of the off-PBX telephone.
The off-PBX telephone is the analog telephone behind the MP-124. Refer to
Section 5 Steps 11-12.
ƒ Set the Trunk Selection field to 1, same as the trunk group configured in
Section 3.1, Step 6.
ƒ 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
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
2346241
CTM; Reviewed:
SPOC 4/19/2007
Application
OPS
Dial
Phone Number
Prefix
- 2346241
-
Trunk
Selection
1
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
Page
1 of
2
Configuration
Set
1
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AudioCodesMP124
Step
5.
Description
Use the change off-pbx-telephone configuration-set n command, where n is a
configuration set number to assign the call treatment options as follows:
ƒ Set the Fast Connect on Origination field to y.
ƒ Leave other fields to their default values.
change off-pbx-telephone configuration-set 1
Page
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1
CONFIGURATION SET: 1
Configuration Set Description:
Calling Number Style:
CDR for Origination:
CDR for Calls to EC500 Destination?
Fast Connect on Origination?
Post Connect Dialing Options:
Cellular Voice Mail Detection:
Barge-in Tone?
Calling Number Verification?
Identity When Bridging:
6.
1
network
phone-number
y
y
dtmf
none
n
y
principal
Repeat Steps 1-4 for each analog endpoint behind the MP-124.
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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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 Avaya SES software and the
license file have already been installed on the server. During the software installation, the
installation script is run from the Linux shell of the server to specify the IP network properties of
the server along with other parameters. For additional information on these installation tasks,
refer to [3].
Step
1.
Description
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 Avaya SES.
Log in with the appropriate credentials and then select the Launch Administration
Web Interface link from the main page as shown below.
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
2.
Description
The Avaya SES Administration main page shown below will be displayed.
CTM; Reviewed:
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©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
3.
Description
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.
CTM; Reviewed:
SPOC 4/19/2007
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©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
Step
4.
Description
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 installation
script during the installation process.
On the Edit System Properties page:
ƒ Enter the SIP Domain name assigned to Avaya SES. This must match the
Authoritative Domain field configured on Avaya Communication Manager
shown in Section 3.1 Steps 3 and 5.
ƒ Enter the License Host field. This is the host name, the fully qualified domain
name, 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 the same as the Avaya SES server.
ƒ After configuring the Edit System Properties page, click the Update button
(not shown).
CTM; Reviewed:
SPOC 4/19/2007
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AudioCodesMP124
Step
5.
Description
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 left
pane and then clicking on the Edit option under the Commands section of the
subsequent page that is displayed (not shown).
ƒ In the Host IP Address field, enter the IP address of Avaya SES.
ƒ In the DB Password field, enter the password that was specified while running
the installation script during the system installation.
ƒ 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 home
and edge server. Thus, the Host Type is configured as home/edge.
ƒ The default values for the other fields may be used.
ƒ Click the Update button at the bottom of the page (not shown).
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Step
6.
Description
From the left pane of the administration web interface, expand the Media Servers
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 the
SIP trunk previously created in Avaya Communication Manager.
On the Add Media Server Interface page, enter the following information:
ƒ A descriptive name in the Media Server Interface Name field (e.g.,
s8500clan).
ƒ In the Host field, select the IP address of the Avaya SES server from the pulldown menu that will serve as the SIP proxy for this media server. Since there is
only one Avaya SES server in this configuration, the Host field is set to the host
shown in Step 5.
ƒ Select TLS (Transport Link Security) for the Link Type. TLS provides
encryption at the transport layer. TLS is the only link protocol that is supported
for communication between Avaya SES and Avaya Communication Manager.
ƒ Enter the IP address of the C-LAN in the SIP Trunk IP Address field. In
configurations that do not use a C-LAN board, the SIP Trunk IP Address
would be the IP address of the Avaya Media Server.
ƒ The default values may be retained for all other fields.
ƒ After completing the Add Media Server Interface page, click the Add button.
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Step
7.
Description
Inbound SIP calls arriving at Avaya SES are routed to the appropriate Avaya
Communication Manager for termination services.
In the compliance test configuration, which has a single Avaya Media Server, calls
originating from SIP endpoints registered to the SES (like the MP-124) will be
automatically routed to the Avaya Media Server. Thus, strictly speaking, the Media
Server Address Maps described below are not necessary. However, these maps were
present during the testing and thus are described below for completeness.
Media Server Address Maps route calls by comparing the Uniform Resource Identifier
(URI) of an incoming INVITE message to the pattern configured in the Media Server
Address Map. If there is a match, the call is routed to the designated Avaya
Communication Manager. The URI usually takes the form of sip: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 Avaya SES server.
The next step shows an example of an address map to match the dialing of an Avaya
Communication Manager FAC. In the compliance test, 3-digit numbers beginning with
digit 8 were configured as FACs.
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Step
7.
cont’d
Description
To configure a Media Server Address Map:
ƒ 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 (not
shown).
ƒ Click on the Map link associated with the appropriate media server to display
the List Media Server Address Map page (not shown).
ƒ Click on the Add Map In New Group link. The page shown below is
displayed. The Host field displays the name of the media server to which this
map applies.
ƒ Enter a descriptive name in the Name field.
ƒ Enter the regular expression to be used for the pattern matching in the Pattern
field. The example below shows the pattern specification for FACs ^sip:8[09]{2}. This expression will match any SIP URI that begins with the text string
sip:8 followed by a 2-digit string. Based on the value of the Host field, these
SIP calls will then be routed to host S8500clan. Appendix A provides a
detailed description of the syntax for address map patterns.
ƒ Click the Add button.
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Step
8.
Description
After submitting the previous form, the List Media Server Address Map page appears
as follows, showing an entry for the newly created Address Map.
Notice that the list includes additional Address Maps. These were created to support
the use of other commonly used dial strings (Abbreviated Dial Codes, etc.). The
procedures used to create these additional Address Maps are similar to those shown in
Step 7, but are not included here.
In addition, a Media Server Contact is created automatically after the first Media Server
Address Map is added. The contact specifies the IP address of the C-LAN board (in
this example, 192.168.10.43) and the transport protocol used to send SIP signaling
messages. The user in the original request URI is substituted for $(user).
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Step
9.
Description
From the left pane of the administration web interface, expand the Users option and
select Add to add a user.
ƒ Specify a value for the 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 ID must be the same as the User Name configured on the MP-124 for an
analog telephone. Refer to Section 5, Step 12.
ƒ Specify a value for the Password field. Enter the same value in the Confirm
Password field. 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 MP-124 for an analog telephone. See Section 5, Step 12.
ƒ The Host field is populated with the Avaya SES Host IP address since there is
only one Avaya SES in the test configuration.
ƒ Enter any descriptive First Name and Last Name.
ƒ Select Add Media Server Extension to add an OPS extension for the user.
Refer to Step 11.
ƒ Click the Add button.
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Step
10.
Description
The following Continue page appears, confirming that the user was added
successfully.
Click Continue.
11.
The Add Media Server Extension page will appear as shown below.
ƒ Enter the Extension, same as the OPS extension configured in Avaya
Communication Manager. Refer to Section 3.2, Step 2.
ƒ Select the appropriate Media Server configured in Step 6.
ƒ Click Add.
12.
Repeat Steps 9-11 to add the users and their associated extensions for additional analog
endpoints behind the MP-124.
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Step
13.
Description
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.
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5. Configure AudioCodes MP-124 Analog VoIP Gateway
The MP-124 is configured via an Internet browser using the embedded web server interface.
Both of the MP-124 gateways shown in Figure 1 are configured using the procedures in this
section. The following steps describe how to configure:
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
ƒ
Step
1.
IP address of the MP-124 and associated 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
Description
Access the MP-124 embedded web server interface, by entering
http://<ip-addr>/ as the URL in an Internet browser, where <ip-addr> is the default IP
address of the MP-124. Refer to [8] for the default IP address.
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Step
2.
Description
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 MP-124, under IP Configuration enter values for the
IP Address, Subnet Mask, and Default Gateway IP Address fields.
To assign basic SIP parameters, under SIP Parameters assign the parameters as shown
below:
ƒ Set the Working with Proxy field to Yes to enable the interworking of the MP124 with Avaya SES proxy server.
ƒ Enter the IP address of the Avaya SES in the Gateway Name, Proxy IP
Address and Proxy Name fields.
ƒ Select Enable Registration to Enable. The AudioCodes will send a
REGISTER request to Avaya SES according to the Authentication Mode
parameter. Refer to Step 5 for Authentication Mode.
ƒ Click the Reset button and click OK in the prompt. The MP-124 applies the
changes and restarts.
After the MP-124 restarts, access the MP-124 embedded web server interface, by
entering http://<ip-addr>/ as the URL in an Internet browser, where <ip-addr> is the
new IP address of the MP-124 configured in this step.
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Step
3.
4.
Description
From the left pane of the web interface, click the Protocol Management option. The
right pane displays configuration options as shown below.
SIP Proxy & Registration
From the right pane of the web interface, click the Protocol Definition Æ Proxy &
Registration option as shown below.
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Step
5.
Description
The basic SIP proxy and registration parameters were configured in Step 2. Scroll
down the SIP Proxy & Registration page to configure additional SIP parameters as
shown below.
ƒ Set the Registration Time to 3600 to specify that the registration for the analog
endpoints behind the MP-124 is valid for one hour.
ƒ Select Send All Invite to Proxy field to Yes so that all the INVITE requests
including those generated as a result of transfer or redirect are sent to Avaya
SES.
ƒ Enter any alphanumeric string in the Cnonce field. This field is used for the
mutual authentication of the server and client.
ƒ Select the Authentication Mode as Per Endpoint. This implies that the
registration and authentication of each audio endpoint behind the MP-124 with
Avaya SES is done separately.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button.
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Step
6.
Description
General Parameters
From the right pane of the web interface, click the Protocol Definition Æ General
Parameters option. Configure general SIP parameters as shown below.
ƒ Select the Enable Early Media field as Enable. If enabled, the MP-124 sends a
183 Session Progress response with Session Description Protocol (SDP)
parameters (instead of 180 Ringing), allowing the media stream to be set up
prior to the answering of the call.
ƒ Select Fax Signaling Method field as T.38 Relay if the T.38 fax transmission is
desired.
ƒ Select the Use “user=phone” in SIP URL field as No.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button (not shown here).
Refer to Step 15 for other T.38 fax settings.
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Step
7.
Description
Coders (Codecs)
From the right pane of the web interface, click the Protocol Definition Æ Coders
option. From the Coders screen, configure up to five preferred coders (and their
attributes) for the MP-124. The first coder is the highest priority coder and is used by
the gateway whenever possible. If the far end gateway (on Avaya Communication
Manager) cannot use the coder assigned as the first coder, the gateway attempts to use
the next coder and so forth.
Configure the coder parameters as shown below.
ƒ From the Coder Name drop-down list, select the appropriate coder. Refer to
Section 3.1, Step 4 for the codec selection and preferred order in Avaya
Communication Manager.
ƒ For G.729 coder, enable the silence suppression by selecting Enable 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 codec
setting in Avaya Communication Manager.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button.
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Step
8.
Description
DTMF and Dialing
From the right pane of the web interface, click the Protocol Definition Æ DTMF &
Dialing option. Use this screen to configure parameters that are associated with DTMF
and dialing. Configure the DTMF and dialing parameters as shown below.
ƒ Enter the maximum number of digits that can be dialed (e.g., 18) in the Max
Digits in Phone Num field.
ƒ From the 1st Tx DTMF Option drop-down list, select RFC 2833 as the
preferred transmit DTMF negotiation method.
ƒ For the RFCC 2833 Payload Type field, enter 101.
ƒ Assign the digit map pattern in the Digit Mapping Rules field. If the digit
string (dialed number) has matched one of the patterns in the digit map, the
MP-124 stops collecting digits and starts to establish a call with the collected
number. The digit map pattern contains up to 52 options separated by a vertical
bar (|). The maximum length of the entire digit pattern is limited to 152
characters.
ƒ
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For example: 2xxxxxx|1xx|91xxxxxxxxxx|8xx|3xx was used in the test
configuration, 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 FACs in
Avaya Communication Manager and 3xx for FNEs in Avaya Communication
Manager.
The default values may be retained for all other fields. After completing this
screen, click the Submit button.
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Step
9.
Description
Advanced Control Parameters
From the right pane of the web interface, click the Advanced Parameters Æ General
Parameters option.
Configure the advanced control parameters as shown below.
ƒ From the Enable Current Disconnect drop down list (under Disconnect and
Answer Supervision), select Enable.
ƒ Set Broken Connection Timeout to desired amount of time an RTP packet
isn’t received, after which the call is disconnected.
ƒ Set the Max Number of Active Calls to the maximum number of channels
available on the gateway. For the tested configuration, it was set to 24.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button (not shown).
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Step
10.
Description
Supplementary Services
From the right pane of the web interface, click the Advanced Parameters Æ
Supplementary Services option.
Enable services such as hold and transfer as shown below.
ƒ From the Enable Caller ID drop down list, select Enable. When enabled, the
calling number and name are sent to the analog endpoints behind the MP-124.
ƒ The default values may be retained for all other fields. Hold, transfer, call
forwarding and call waiting are enabled by default.
ƒ After completing this screen, click the Submit button.
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Step
11.
Description
Endpoint Phone Numbers
From 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 MP-124, to
which the analog endpoints are connected.
ƒ Channel is a port number on the back of the MP-124. To enable a channel,
enter a port number.
ƒ 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, Steps 9-14 for how to configure related
telephone extensions in Avaya Communication Manager and Avaya SES.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button (not shown).
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Step
12.
Description
Endpoint Authentication
From the right pane of the web interface, click the Endpoint Settings Æ
Authentication option.
The Authentication table defines a user name and password combination for
authentication for each MP-124 port as shown below.
ƒ Enter a value for the User Name and Password fields for each port. This is
used for registration and authentication with Avaya SES. Make sure that these
values are the same as configured using Section 4, Step 9.
ƒ After completing this screen, click the Submit button (not shown).
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Step
13.
Description
Enable Caller ID
From the right pane of the web interface, click the Endpoint Settings Æ Generate
Caller ID to Tel option.
The Generate Caller ID to Tel table is used to enable or disable (per port) the caller
ID generation and detection as shown below.
ƒ From the Caller ID drop down list for each port, select Enable or Disable. In
the tested configuration the caller ID was enabled for 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 ring cycle. The
caller ID type defaults to BellCore (see Step 10). If set to other call types, the
caller ID display and behavior may be different.
ƒ The default values may be retained for all other fields.
ƒ After completing this screen, click the Submit button (not shown).
14.
From the left pane of the web interface, click the Advanced Configuration option.
The right pane displays configuration options as shown below.
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Step
15.
Description
T.38 Fax
From the right pane of the web interface, click the Media Settings Æ
Fax/Modem/CID Settings option and configure the fax option as shown below.
ƒ From the Fax Transport Mode drop down list, select T.38 Relay if T.38 fax is
desired.
ƒ Use the default values for all other fields.
ƒ After completing this screen, click the Submit button.
Refer to Step 6 for other T.38 fax settings.
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Step
16.
17.
Description
For proper modem interoperability, an additional parameter must be changed in the ini
file. From a web browser, enter <ip_address>/AdminPage where <ip_address> is the
IP address of the MP-124. The main page will appear as shown below.
Select 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 Apply New Value.
A value of 0 in the payload type indicates a G.711MU call. By default, the MP-124
uses a proprietary value for the payload type of modem calls so these calls can be
distinguished from other G.711MU calls.
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6. Interoperability Compliance Testing
This section describes the compliance testing used to verify the interoperability between the
AudioCodes MP-124 Analog VoIP Gateway, Avaya SIP Enablement Services 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 to/from the telephones connected to the MP-124 at
the enterprise sites using various codec settings and exercising common PBX features. The calls
were made to/from the PSTN and within the enterprise site.
6.2. Test Results
The MP-124 successfully passed compliance testing. The following features and functionality
were verified.
ƒ Calls to/from the PSTN
ƒ Calls between the MP-124 analog telephones
ƒ Calls between the MP-124 analog telephones and the Avaya 4600 Series IP Telephones
(H.323)
ƒ G.711MU and G.729AB codec support
ƒ Proper recognition of DTMF transmissions
ƒ MP-124 support for hold, transfer (attended and unattended), and call waiting
ƒ Proper operation of voicemail with message waiting indicators (MWI); for the analog
phones, MWI was provided via stutter dial tone
ƒ Call forwarding provided by Avaya Communication Manager
ƒ Conferencing (Avaya H.323 IP telephone initiates a conference that includes an MP-124
analog endpoint)
ƒ A subset of Avaya Communication Manager features (e.g., hunt group, call pickup group,
call forward, and send all calls) involving the analog telephones behind the MP-124
ƒ T.38 fax support
ƒ Modem calls
ƒ Proper system recovery after a MP-124 restart
The following observations were made during the compliance test:
ƒ To support T.38 fax with the MP-124, a post-GA release (R013x.01.2.635.0) of Avaya
Communication Manager was required.
ƒ The MP-124 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. This feature was tested with the GA release.
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7. Verification Steps
This section provides verification steps that may be performed to verify that the solution
described in these Application Notes is configured properly.
ƒ
ƒ
ƒ
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ƒ
ƒ
ƒ
ƒ
ƒ
Verify that the SIP trunk group is in-service. To do this, use the status trunk n
command, where n 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 signalinggroup n command, where n is the number of the signaling group to be verified.
Verify that an analog telephone behind the MP-124 can register with Avaya SES. 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 MP-124.
Verify that a call can be placed between an analog telephone behind the MP-124 and a
telephone in the PSTN.
Verify that a call can be placed between an analog telephone behind the MP-124 and an
Avaya 4600 Series IP Telephone (H.323).
Verify that a T.38 fax can be transmitted between a fax machine behind the MP-124 and
a fax machine in the PSTN.
Verify that a data modem call can be placed between a data modem behind the MP-124
and a data modem in the PSTN.
Verify that the transfer and hold features can be successfully activated from the analog
telephones behind the 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
pickup group, call forward, and send all calls) that involve the analog telephones behind
the MP-124.
8. Support
Technical support for the AudioCodes MP-124 Analog VoIP Gateway can be obtained from
AudioCodes. See the Support link at www.audiocodes.com for contact information.
9. Conclusion
The analog telephones, modems and fax machines behind the AudioCodes MP-124 Analog VoIP
Gateway can successfully register to Avaya SIP Enablement Services, place/receive calls and
utilize telephony features of Avaya Communication Manager.
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10. Additional References
The 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 Number 555-245-206.
[5] 4600 Series IP Telephone Release 2.4 LAN Administrator Guide, April 2006, Issue 2.3,
Document Number 555-233-507.
[6] Avaya Extension to Cellular User’s Guide, February 2006, Issue 9, Document Number
210-100-700.
[7] Avaya Extension to Cellular and OPS Installation and Administration Guide, June 2005,
Issue 9, Document Number 210-100-500.
The following MP-124 product documentation is available from AudioCodes. Visit
http://www.audiocodes.com for company and product information.
[8] AudioCodes MediaPack MP-124 and MP-11x User’s Manual 4.8.
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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:
ƒ Normal text characters and numbers match themselves.
ƒ 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:17325551638@20.1.1.54:5060;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 devconnect@avaya.com.
CTM; Reviewed:
SPOC 4/19/2007
Solution & Interoperability Test Lab Application Notes
©2007 Avaya Inc. All rights reserved.
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AudioCodesMP124
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