Asm CM Cucme

Avaya Solution & Interoperability Test Lab
Configuring SIP Trunks among Avaya Aura Session

Manager, Avaya Aura Communication Manager, and
Cisco Unified Communications Manager Express Issue 1.0
Abstract
These Application Notes present a sample configuration for a network that uses Avaya Aura
Session Manager to connect Avaya Aura Communication Manager and Cisco Unified
Communications Manager Express using SIP trunks.
For the sample configuration, Avaya Aura Session Manager runs on an Avaya S8510
Server, Avaya Aura Communication Manager runs on an Avaya S8300 Server with Avaya
G430 Media Gateway, and Cisco Unified Communications Manager Express runs on a Cisco
3825 Integrated Services Router (ISR). The results in these Application Notes should be
applicable to other Avaya servers and media gateways that support Avaya Aura
Communication Manager.
AM; Reviewed:
SPOC 08/10/2009
Solution & Interoperability Test Lab Application Notes

2009 Avaya Inc. All Rights Reserved.
1 of 48
ASM-CM-CUCME
1 Introduction
These Application Notes present a sample configuration for a network that uses Avaya Aura
Session Manager to connect Avaya Aura Communication Manager and Cisco Unified
Communications Manager Express (Cisco UCME) using SIP trunks.
As shown in Figure 1, the Avaya 9630 IP Telephone (H.323) and 6408D+ Digital Telephone are
supported by Avaya Aura Communication Manager. The Cisco 7965G IP Telephone (SIP)
and the Cisco 7975G IP Telephone (SCCP) are supported by the Cisco UCME. SIP trunks are
used to connect these two systems to Avaya Aura Session Manager, using its SM-100
(Security Module) network interface. All inter-system calls are carried over these SIP trunks.
Avaya Aura Session Manager can support flexible inter-system call routing based on dialed
number, calling number and system location, and can also provide protocol adaptation to allow
multi-vendor systems to interoperate. It is managed by a separate Avaya Aura System
Manager, which can manage multiple Avaya Aura Session Managers by communicating with
their management network interfaces. Configurations supporting SIP telephones still require
Avaya SIP Enablement Services, and are not addressed in these application notes.
Figure 1 Sample Configuration

AM; Reviewed:
SPOC 08/10/2009

2 of 48
ASM-CM-CUCME
For the sample configuration, Avaya Aura Session Manager runs on an Avaya S8510 Server,
Avaya Aura Communication Manager runs on an Avaya S8300 Server with Avaya G430
Media Gateway, and Cisco Unified Communications Manager Express runs on Cisco 3825
Integrated Services Router (ISR). The results in these Application Notes should be applicable to
other Avaya Aura servers and Media Gateways.
A five digit Uniform Dial Plan (UDP) is used for dialing between systems. Unique extension
ranges are associated with Avaya Aura Communication Manager 5.2 (143xx) and Cisco
UCME (777xx).
These Application Notes will focus on the configuration of the SIP trunks and call routing.
Detailed administration of the endpoint telephones will not be described (see the appropriate
documentation listed in Section 8).
AM; Reviewed:
SPOC 08/10/2009

3 of 48
ASM-CM-CUCME
2 Equipment and Software Validated

The following equipment and software were used for the sample configuration provided:
DEVICE DESCRIPTION
Avaya Aura Communication Manager
- Running on an Avaya S8300 Server with an
Avaya G430 Media Gateway
Avaya Aura System Manager
- Running on an Avaya S8510 Server
Avaya Aura Session Manager
- Running on an Avaya S8510 Server
Avaya 9630 IP Telephone (H.323)
Avaya 6402D Digital Telephone
Cisco Unified Communications Manager Express
- Running on a Cisco 3825 ISR
Cisco 7965G Unified IP Phone (SIP)
Cisco 7975G Unified IP Phone (SCCP)
VERSION TESTED
Communication Manager 5.2
(R015x.02.0.947.3.) with
SP1 (02.0.947.3-17294)
1.0
1.1 with SP1
3.0
7.1
IOS 12.4(24)T1
8.4.2S
8.4.2S
3 Configure Avaya Aura Communication Manager

This section provides the procedures for configuring Avaya Aura Communication Manager.
The procedures include the following areas:
Verify Avaya Aura Communication Manager license

Administer system parameters features
Administer IP node names
Administer IP codec set and network region
Administer SIP trunk group and signaling group
Administer SIP trunk group members and route patterns
Administer location and public unknown numbering
Administer uniform dial plan and AAR analysis
Some administration screens have been abbreviated for clarity.
AM; Reviewed:
SPOC 08/10/2009

4 of 48
ASM-CM-CUCME
3.1
Verify Avaya Aura Communication Manager License
Log into the System Access Terminal (SAT) to verify that the Avaya Aura Communication
Manager license has proper permissions for features illustrated in these Application Notes. Use
the display system-parameters customer-options command. Navigate to Page 2, and verify
that there is sufficient remaining capacity for SIP trunks by comparing the Maximum
Administered SIP Trunks field value with the corresponding value in the USED column. The
difference between the two values needs to be greater than or equal to the desired number of
simultaneous SIP trunk connections.
The license file installed on the system controls the maximum permitted. If there is insufficient
capacity or a required feature is not enabled, 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:
3.2
800
18000
0
0
0
0
0
0
800
2 of
10
USED
200
2
0
0
0
0
0
0
47
Configure System Parameters Features
Use the change system-parameters features command to allow for trunk-to-trunk transfers.
Submit the change.
This feature is needed to be able to transfer an incoming/outgoing call from/to the remote switch
back out to the same or another switch For simplicity, the Trunk-to-Trunk Transfer field was
set to all to enable all trunk-to-trunk transfers on a system wide basis. Note that this feature
poses significant security risk, and must be used with caution. For alternatives, the trunk-totrunk feature can be implemented using Class Of Restriction or Class Of Service levels. Refer to
the appropriate documentation in Section 8 for more details.
change system-parameters features
Page
1 of
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? y
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
DID/Tie/ISDN/SIP Intercept Treatment: attd
Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred
AM; Reviewed:
SPOC 08/10/2009

18
5 of 48
ASM-CM-CUCME
3.3
Configure IP Node Names
Use the change node-names ip command to add entries for the procr and Avaya Aura
Session Manager that will be used for connectivity. In the sample network, procr and
172.28.43.5 are entered as Name and IP Address for the Avaya Aura Communication
Manager running on the Avaya S8300 Server. In addition, ASM and 10.1.2.170 are entered
for Avaya Aura Session Manager Security Module (SM-100) interface. The actual node names
and IP addresses may vary. Submit these changes.
change node-names ip
Page
1 of
IP NODE NAMES
Name
ASM
default
msgserver
procr
3.4
IP Address
10.1.2.170
0.0.0.0
172.28.43.9
172.28.43.5
Configure IP Codec Sets and Network Regions
Configure the IP codec set to use for calls to the Cisco UCME. Use the change ip-codec-set n
command, where n is an existing codec set number to be used for interoperability. Enter the
desired audio codec type in the Audio Codec field. Retain the default values for the remaining
fields and submit these changes.
In addition to the G.711MU codec shown below, G.729 and G.729AB have also been
verified to be interoperable with Cisco UCME via SIP trunks.
change ip-codec-set 1
Page
1 of
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
3:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
In the test configuration, network region 1 was used for calls to the Cisco UCME via Avaya
Aura Session Manager. Use the change ip-network-region 1 command to configure this
network region. For the Authoritative Domain field, enter the SIP domain name configured for
this enterprise network (See Section 4.1). This value is used to populate the SIP domain in the
From header of SIP INVITE messages for outbound calls. It also must match the SIP domain in
the request URI of incoming INVITEs from other systems. Enter a descriptive Name. For the
Codec Set field, enter the corresponding audio codec set configured above in this section.
Enable the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio. These
settings will enable direct media between Avaya IP telephones and the far end. Retain the
default values for the remaining fields, and submit these changes.
AM; Reviewed:
SPOC 08/10/2009

6 of 48
ASM-CM-CUCME
change ip-network-region 1
Page
1 of
19
IP NETWORK REGION
Region: 1
Location:
Authoritative Domain: avaya.com
Name: ASM to Cisco UCME
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio:
Codec Set: 1
Inter-region IP-IP Direct Audio:
UDP Port Min: 2048
IP Audio Hairpinning?
UDP Port Max: 10001
DIFFSERV/TOS PARAMETERS
RTCP Reporting Enabled?
Call Control PHB Value: 46
RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46
Use Default Server Parameters?
Video PHB Value: 26
3.5
yes
yes
n
y
y
Configure SIP Signaling Group and Trunk Group
3.5.1 SIP Signaling Group

In the test configuration, trunk group 25 and signaling group 25 were used to reach Avaya
Aura Session Manager. Use the add signaling-group n command, where n is an available
signaling group number. Enter the following values for the specified fields, and retain the
default values for all remaining fields. Submit these changes.
Group Type:
Transport Method:
Near-end Node Name:
Far-end Node Name:
Near-end Listen Port:
Far-end Listen Port:
Far-end Network Region:
Far-end Domain:
DTMF over IP:
sip
tls
Processor card node name procr from Section 3.3.
Avaya Aura Session Manager node name from Section 3.3.
5061
5061
Avaya network region number 1 from Section 3.4.
Fields is left blank so that the signaling group accepts any
authoritative domain.
rtp-payload
add signaling-group 25
Page
1 of
SIGNALING GROUP
Group Number: 25
Group Type: sip

Transport Method: tls
IMS Enabled? n
Near-end Node Name: procr
Near-end Listen Port: 5061
Far-end Node Name: ASM

Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain:
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload
Enable Layer 3 Test? n
Session Establishment Timer(min): 3
AM; Reviewed:
SPOC 08/10/2009
Direct IP-IP Audio Connections?

IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):

y
n
n
6
7 of 48
ASM-CM-CUCME
3.5.2 SIP Trunk Group

Use the add trunk-group n command, where n is an available trunk group number. Enter the
following values for the specified fields, and retain the default values for the remaining fields.
Group Type:
Group Name:
TAC:
Service Type:
Number of Members:
sip
A descriptive name.
An available trunk access code.
tie
The number of SIP trunks to be allocated to calls routed to Avaya
Aura Session Manager (must be within the limits of the total
trunks configure in Section 3.1).
add trunk-group 25
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
Group Type: sip

CDR Reports: y
COR: 1
TN: 1
TAC: 125
Outgoing Display? y
Night Service:
25
To ASM
two-way
n
0
tie
Auth Code? n
Signaling Group: 25
Number of Members: 10
Navigate to Page 3, and enter public for the Numbering Format field as shown below. Use
default values for all other fields.
add trunk-group 25
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: public

UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Navigate to Page 4, and enter 101 for the Telephone Event Payload Type field as shown
below. Use default values for all other fields. Submit these changes.
add trunk-group 25
Page
4 of
21
PROTOCOL VARIATIONS
Mark Users as Phone?
Prepend '+' to Calling Number?
Send Transferring Party Information?
Network Call Redirection?
Send Diversion Header?
Support Request History?
Telephone Event Payload Type:
AM; Reviewed:
SPOC 08/10/2009
n
n
n
n
n
y
101

8 of 48
ASM-CM-CUCME
3.6
Configure Route Pattern
Configure a route pattern to correspond to the newly added SIP trunk group. Use the change
route-pattern n command, where n is an available route pattern. Enter the following values
for the specified fields, and retain the default values for the remaining fields. Submit these
changes.
Pattern Name: A descriptive name.
Grp No:
The trunk group number from Section 3.5.2.
FRL:
Enter a level that allows access to this trunk, with 0 being least restrictive.
change route-pattern 25
Pattern Number: 25 Pattern Name: To ASM
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 25
0
2:
3:
4:
5:
6:
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
1: y y y y y n
AM; Reviewed:
SPOC 08/10/2009
Page
1 of
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
ITC BCIE Service/Feature PARM
rest
No. Numbering LAR

Dgts Format
Subaddress
none

9 of 48
ASM-CM-CUCME
3.7
Configure Location and Public Unknown Numbering
Use the change locations command to assign the SIP route pattern for Avaya SIP endpoints to
a location corresponding to the Main site. Add an entry for the Main site if one does not exist
already, enter the following values for the specified fields, and retain default values for the
remaining fields. Submit these changes.
Name:
Timezone:
Rule:
Proxy Sel. Rte. Pat.:
A descriptive name to denote the Main site.

An appropriate timezone offset.
An appropriate daylight savings rule.
The Avaya route pattern number from Section 3.6.
change locations
Page
1 of
LOCATIONS
ARS Prefix 1 Required For 10-Digit NANP Calls? y
Loc Name
No
1: Main
Timezone Rule
Offset
+ 00:00
0
NPA
Proxy Sel
Rte Pat
25
Use the change public-unknown-numbering 0 command, to define the calling party number to
be sent to Cisco UCME. Add an entry for the trunk group defined in Section 3.5.2 to reach
Cisco endpoints. In the example shown below, all calls originating from a 5-digit extension
beginning with 777 and routed to trunk group 25 will result in a 5-digit calling number. The
calling party number will be in the SIP From header. Submit these changes.
change public-unknown-numbering 0
Page
1 of
2
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 2
5 777
25
5
Maximum Entries: 9999
AM; Reviewed:
SPOC 08/10/2009

10 of 48
ASM-CM-CUCME
3.8
Administer Uniform Dial Plan and AAR Analysis
This section provides sample Automatic Alternate Routing (AAR) used for routing calls with
dialed digits 777xx to Cisco UCME. Note that other methods of routing may be used. Use the
change uniform-dialplan 0 command, and add an entry to specify use of AAR for routing of
digits 777xx. Enter the following values for the specified fields, and retain the default values for
the remaining fields. Submit these changes.
Matching Pattern:
Len:
Del:
Net:
Dialed prefix digits to match on, in this case 777.

Length of the full dialed number.
Number of digits to delete.
aar
change uniform-dialplan 0
UNIFORM DIAL PLAN TABLE
Page
1 of
Percent Full: 0
Matching
Pattern
777
Insert
Digits
Len Del
5
0
Node
Net Conv Num
aar n
Use the change aar analysis 0 command, and add an entry to specify how to route the calls to
777xx. Enter the following values for the specified fields, and retain the default values for the
remaining fields. Submit these changes.
Dialed String:
Total Min:
Total Max:
Route Pattern:
Call Type:
Dialed prefix digits to match on, in this case 777.

Minimum number of digits.
Maximum number of digits.
The route pattern number from Section 3.6.
aar
change aar analysis 0
Page
AAR DIGIT ANALYSIS TABLE
Location: all
Dialed
String
777
3.9
Total
Min Max
5
5
Route
Pattern
25
Call
Type
aar
1 of
Percent Full:
Node
Num
2
1
ANI
Reqd
n
Save Translations
Configuration of Avaya Aura Communication Manager is complete. Use the save

Translations command to save these changes.
AM; Reviewed:
SPOC 08/10/2009

11 of 48
ASM-CM-CUCME
4 Configure Avaya Aura Session Manager

This section provides the procedures for configuring Avaya Aura Session Manager. The
procedures include adding the following items:
SIP domain
Logical/physical Locations that can be occupied by SIP Entities
Adaptations for digit conversion and Cisco SIP header manipulation
SIP Entities corresponding to the SIP telephony systems and Avaya Aura Session
Manager
Entity Links, which define the SIP trunk parameters used by Avaya Aura Session
Manager when routing calls to/from SIP Entities
Time Ranges during which routing policies are active
Routing Policies, which control call routing between the SIP Entities
Dial Patterns, which govern to which SIP Entity a call is routed
Session Manager, corresponding to the Avaya Aura Session Manager Server to be managed
by Avaya Aura System Manager.
Configuration is accomplished by accessing the browser-based GUI of Avaya Aura System

Manager, using the URL http://<ip-address>/IMSM, where <ip-address> is the IP address of
Avaya Aura System Manager. Log in with the appropriate credentials and accept the
Copyright Notice.
AM; Reviewed:
SPOC 08/10/2009

12 of 48
ASM-CM-CUCME
The menu shown below is displayed. Expand the Network Routing Policy Link on the left side
as shown. The sub-menus displayed in the left column below will be used to configure all but
the last of the items mentioned earlier (Sections 4.1 through 4.8).
AM; Reviewed:
SPOC 08/10/2009

13 of 48
ASM-CM-CUCME
4.1
Specify SIP Domain
Add the SIP domain for which the communications infrastructure will be authoritative. Do this
by selecting SIP Domains on the left and clicking the New button on the right. The following
screen will then be shown. Fill in the following:
Name:
Notes:
The authoritative domain name (e.g., avaya.com)

Descriptive text (optional).
Click Commit.
Since the sample configuration does not deal with any other domains, no additional domains
need to be added.
AM; Reviewed:
SPOC 08/10/2009

14 of 48
ASM-CM-CUCME
4.2
Add Locations
Locations can be used to identify logical and/or physical locations where SIP Entities reside, for
purposes of bandwidth management. Locations are added for the Avaya and the Cisco
environments. To add a location, select Locations on the left and click on the New button on the
right. The following screen will then be shown. Fill in the following:
Under General:
Name:
Notes:
A descriptive name.
Under Location Pattern:

IP Address Pattern:
Notes:
A pattern used to logically identify the location.

The screen below shows the addition of the Lincroft location, which includes Avaya Aura
Communication Manager and Avaya Aura Session Manager. Click Commit to save each
Location definition.
AM; Reviewed:
SPOC 08/10/2009

15 of 48
ASM-CM-CUCME
The following screen shows the addition of a second location based on the subnet used by Cisco
UCME.
The fields under General can be filled in to specify bandwidth management parameters between
Avaya Aura Session Manager and this location. These were not used in the sample
configuration, and reflect default values. Note also that although not implemented in the sample
configuration, routing policies can be defined based on location.
AM; Reviewed:
SPOC 08/10/2009

16 of 48
ASM-CM-CUCME
4.3
Add Adaptations
Two Adaptations need to be created: One for calls from/to Avaya Aura Communication
Manager DigitConversionAdapter and the other for calls from/to Cisco UCME CiscoAdapter.
The DigitConversionAdapter will adapt SIP request and SIP response messages. It uses the
following pieces of information to perform digit adaptation on various SIP headers:
Adaptation direction (incoming/ingress or outgoing/egress)
Matching digit pattern and corresponding digits to remove/insert
Domain name change for source components and destination components
The origination/source type headers are:
P-Asserted-Identity
History-Info (calling portion)
Contact (in 3xx response)
The destination type headers are:
Request-URI
Message Account in NOTIFY (message-summary body)
Refer-To (in REFER message)
The CiscoAdapter provides two basic header manipulations: converting between Diversion
and History-Info headers and converting between P-Asserted-Id and Remote-Party-Id headers.
The Diversion and Remote-Party-Id headers have not been accepted by the IETF. They are
replaced by History-Info and P-Asserted-Identity respectively, but are still used in the Cisco
products. The Cisco Adapter will also perform all the conversions available by the
DigitConversionAdapter.
AM; Reviewed:
SPOC 08/10/2009

17 of 48
ASM-CM-CUCME
For the Avaya Aura Communication Manager adaptation, enter the following information.
An informative name for the adaptation (e.g., Avaya-G430)
Enter DigitConversionAdapter avaya.com, where the domain
name replaces the domain name in the Request-URI, History-Info
header (calling part), and Notify message-summary body.
Digit Conversion for Incoming Calls
Matching Pattern 143 with a minimum and maximum of 5 digits
long, which is the dial pattern for station registered with Avaya
Aura Communication Manager.
Digit Conversion for Outgoing Calls
Matching Patterns 777 with a minimum and maximum of 5 digits
long, which is the dial pattern for station registered with Cisco
UCME.
Name
Adaptation Module
AM; Reviewed:
SPOC 08/10/2009

18 of 48
ASM-CM-CUCME
For the Cisco UCME adaptation, enter the following information.

CiscoUCME, an informative name for the adaptation
Enter CiscoAdapter avaya.com, where the domain name replaces
the domain name in the Request-URI, History-Info header (calling
part), and Notify message-summary body.
Digit Conversion for Incoming Calls
long, which is the dial pattern for station registered with Cisco
UCME.
Digit Conversion for Outgoing Calls
long, which is the dial pattern for station registered with Avaya
Aura Communication Manager.
Name
Adaptation Module
AM; Reviewed:
SPOC 08/10/2009

19 of 48
ASM-CM-CUCME
4.4
Add SIP Entities
A SIP Entity must be added for Avaya Aura Session Manager and for each SIP-based
telephony system supported by it using SIP trunks. In the sample configuration a SIP Entity is
added for the ASM, the Avaya S8300 server, and the Cisco UCME. To add a SIP Entity, select
SIP Entities on the left and click on the New button on the right. The screen is displayed as
shown on the next page. Fill in the following:
Under General:
Name:
FQDN or IP Address:
Type:
Adaptation:
Location:
Time Zone:
A descriptive name.
IP address of the ASM or the signaling interface on
the telephony system.
Session Manager for Avaya Aura Session Manager,
CM for Avaya Aura Communication Manager, and
Other for Cisco UCME.
Select appropriate adaptation (Note: Not needed for SM1).
Select one of the locations defined in Section 4.2.
Time zone for this location.
Under Port for the Avaya Aura Session Manager Entity, click Add, and then edit the fields in
the resulting new row as shown below:
Port:
Port number on which the system listens for SIP requests.
Protocol:
Transport protocol to be used to send SIP requests.
Default Domain
The domain used for the enterprise (e.g., avaya.com).
Defaults can be used for the remaining fields. Click Commit to save each SIP Entity definition.
AM; Reviewed:
SPOC 08/10/2009

20 of 48
ASM-CM-CUCME
The following screen shows addition of Avaya Aura Session Manager. The IP address used is
that of the SM-100 Security Module.
AM; Reviewed:
SPOC 08/10/2009

21 of 48
ASM-CM-CUCME
The following screen shows addition of Avaya Aura Communication Manager. The IP
address used is that of the Avaya S8300 server.
AM; Reviewed:
SPOC 08/10/2009

22 of 48
ASM-CM-CUCME
The following screen shows addition of Cisco UCME. The IP address used is that of the Cisco
UCME ethernet interface.
AM; Reviewed:
SPOC 08/10/2009

23 of 48
ASM-CM-CUCME
4.5
Add Entity Links
A SIP trunk between Avaya Aura Session Manager and a telephony system is described by an
Entity link. To add an Entity Link, select Entity Links on the left and click on the New button
on the right. Fill in the following fields in the new row that is displayed:
Name:
SIP Entity 1:
Port:
SIP Entity 2:
Port:
Trusted:
A descriptive name.
Select the Avaya Aura Session Manager.
Port number to which the other system sends SIP requests
Select the name of the other system.
Port number on which the other system receives SIP
requests
Check this box. Note: If this box is not checked, calls
from the associated SIP Entity specified in Section 4.4
will be denied.
Click Commit to save each Entity Link definition. The following screen shows the result of
adding the two Entity Links for Avaya Aura Communication Manager and Cisco UCME.
AM; Reviewed:
SPOC 08/10/2009

24 of 48
ASM-CM-CUCME
4.6
Add Time Ranges
Before adding routing policies (see Section 4.7), time ranges must be defined during which the
policies will be active. In the sample configuration, one policy was defined that would allow
routing to occur at anytime. To add this time range, select Time Ranges on the left and click on
the New button on the right. Fill in the following:
A descriptive name (e.g., Anytime).

Check the box under each of these headings
Enter 00:00.
Enter 23:59
Name:
Mo through Su
Start Time
End Time
Click Commit to save this time range.
AM; Reviewed:
SPOC 08/10/2009

25 of 48
ASM-CM-CUCME
4.7
Add Routing Policies
Routing policies describe the conditions under which calls will be routed to the SIP Entities
specified in Section 4.4. Two routing policies must be added one for Avaya Aura
Communication Manager and one for Cisco UCME. To add a routing policy, select Routing
Policies on the left and click on the New button on the right. The screen shown on the next page
is displayed. Fill in the following:
Under General:
Enter a descriptive name in Name.
Under SIP Entity as Destination:
Click Select, and then select the appropriate SIP entity to which this routing policy
applies.
Under Time of Day:
Click Add, and select the time range configured in Section 4.6.
Defaults can be used for the remaining fields. Click Commit to save each Routing Policy
definition. The following screens show the Routing Policy for Avaya Aura Communication
Manager and one for Cisco UCME.
AM; Reviewed:
SPOC 08/10/2009

26 of 48
ASM-CM-CUCME
AM; Reviewed:
SPOC 08/10/2009

27 of 48
ASM-CM-CUCME
AM; Reviewed:
SPOC 08/10/2009

28 of 48
ASM-CM-CUCME
4.8
Add Dial Patterns
Dial patterns must be defined that will direct calls to the appropriate SIP Entity. In the sample
configuration, 5-digit extensions beginning with 143 reside on Avaya Aura Communication
Manager, and 5-digit extensions beginning with 777 reside on Cisco UCME. To add a dial
pattern, select Dial Patterns on the left and click on the New button on the right. Fill in the
following, as shown in the screen later in this section, which corresponds to the dial pattern for
routing calls to Avaya Aura Communication Manager:
Under General:
Pattern:
Min
Max
Notes
Dialed number or prefix.

Minimum length of dialed number.
Maximum length of dialed number.
Comment on purpose of dial pattern.
Under Originating Locations and Routing Policies:

Click Add, and then select the appropriate location and routing policy from the list.
Default values can be used for the remaining fields. Click Commit to save this dial pattern. The
following screens show the dial pattern definitions for Avaya Aura Communication Manager
and Cisco UCME.
AM; Reviewed:
SPOC 08/10/2009

29 of 48
ASM-CM-CUCME
AM; Reviewed:
SPOC 08/10/2009

30 of 48
ASM-CM-CUCME
AM; Reviewed:
SPOC 08/10/2009

31 of 48
ASM-CM-CUCME
4.9
Add Session Manager
To complete the configuration, adding the Session Manager will provide the linkage between
Avaya Aura System Manager and Avaya Aura Session Manager. Expand the Session
Manager menu on the left and select Session Manager Administration. Then click Add, and
fill in the fields as described below and shown in the following screen:
Under Identity:
SIP Entity Name:
Select the name of the SIP Entity added for Avaya Aura
Session Manager
Description:
Descriptive comment (optional)
Management Access Point Host Name/IP
Enter the IP address of the Avaya Aura Session
Manager management interface.
Under Security Module:
Network Mask:
Enter the network mask corresponding to the IP address of
Avaya Aura Session Manager
Default Gateway:
Enter the IP address of the default gateway for Avaya
Aura Session Manager
Use default values for the remaining fields. Click Save to add this Session Manager.
AM; Reviewed:
SPOC 08/10/2009

32 of 48
ASM-CM-CUCME
5 Configure Cisco Unified Communications Manager

Express
Cisco Unified Communications Manager Express (Cisco UCME) is a call-processing application
in Cisco IOS software that enables Cisco routers to deliver key-system or hybrid Private Branch
Exchange (PBX) functionality for enterprise branch offices or small businesses. It supports
H.323 and SIP trunk operation to other IP PBX systems.
This section illustrates the relevant Cisco UCME configuration for SIP trunking to Avaya
Aura Communication Manager via Avaya Aura Session Manager. A VoIP dial peer trunk
is configured in the Cisco UCME to connect to the Avaya Aura Session Manager for
communication with Avaya Aura Communication Manager.
With the Cisco IOS 12.4(24)T1 used in this configuration, Cisco 7965G SIP and 7975G SSCP
telephones require the following firmware to work with the Cisco UCME.
Cisco 7965G SIP Telephone:
SIP45.8-2-2S.loads
term45.default.loads
apps45.8-4-1-23.sbn
cnu45.8-4-1-23.sbn
cvm45sip.8-4-1-23.sbn
dsp45.8-4-1-23.sbn
jar45sip.8-4-1-23.sbn
Cisco 7975G SCCP Telephone:
SCCP75.8-4-1-23.loads
jar75sccp.8-4-1-23.sbn
cvm75sccp.8-4-1-23.sbn
apps75.8-4-1-23.sbn
cnu75.8-4-1-23.sbn
dsp75.8-4-1-23.sbn
This section focuses on the VoIP related configuration (in bold) on the Cisco 3825 router.
version 12.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
AM; Reviewed:
SPOC 08/10/2009

33 of 48
ASM-CM-CUCME
service sequence-numbers
!
hostname UCME-3825
!
boot-start-marker
boot system flash c3825-ipvoicek9-mz.124-24.T1.bin
--- Boot image
boot-end-marker
!
!card type command needed for slot 1
security authentication failure rate 3 log
security passwords min-length 6
logging message-counter syslog
logging buffered 51200
logging console critical
enable secret 5 $1$vrfA$TvozCsgK1j/m.gohuDw7Q1
!
no aaa new-model
clock timezone edt -5
clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00
clock calendar-valid
no network-clock-participate slot 1
!
dot11 syslog
no ip source-route
ip cef
!
no ip dhcp use vrf connected
ip dhcp excluded-address 192.45.131.1 192.45.131.9
ip dhcp excluded-address 192.45.131.100 192.45.131.254
!
ip dhcp pool ucme
--- DHCP server configuration
import all
network 192.45.131.0 255.255.255.0
--- Network/subnet configuration
default-router 192.45.131.2
--- Default router configuration
option 150 ip 192.45.131.1
--- Use option 150 to set UCME as TFTP server
!
no ip bootp server
no ip domain lookup
ip domain name interoplab.local
ip name-server 192.45.132.182
no ipv6 cef
multilink bundle-name authenticated
!
voice-card 0
--- Enable card to share DSP resources
dspfarm
AM; Reviewed:
SPOC 08/10/2009

34 of 48
ASM-CM-CUCME
dsp services dspfarm

!
voice call carrier capacity active
!
voice service voip
--- Voice Class and Service VoIP configuration
allow-connections sip to sip
--- Enable B2BUA and allow SIP-SIP connections
no supplementary-service sip moved-temporarily --- Disable sending 302
no supplementary-service sip refer
--- Disable sending REFER
redirect ip2ip
--- Allow SIP calls to be hairpinned through UCME
sip
registrar server
--- Enable SIP Registrar service for SIP endpoints
g729 annexb-all
--- Enable UCME to accept all G729 codec flavors
!
voice class codec 1
--- Voice Class Codec configuration
codec preference 1 g711ulaw
--- Configure G.711 u-law as first preferred codec
!
voice register global
--- SIP global settings for SIP phone registration
mode cme
--- Enable mode for provisioning SIP phones on CM
source-address 192.45.131.1 port 5060
--- Enable UCME router to receive SIP messages
max-dn 50
--- Define max dn number supported on UCME
max-pool 20
--- Limited the # of SIP phones supported by UCME.
load 7965 SIP45.8-4-2S
--- Associate a 7965 phone type with a phone firmware file
timezone 12
--- Configure timezone
dialplan-pattern 2 777.. extension-length 5 --- Define dialplan-pattern for the Cisco UCME stations
dialplan-pattern 4 14... extension-length 5 --- Define dialplan-pattern for the Avaya stations
external-ring bellcore-dr3
--- Define external voicemail access number
voicemail 20000
--- Define voicemail access number
create profile
--- Create a profile on UCME
!
voice register dn 2
--- SIP phone directory number settings
number 77702
--- Define directory number (extension)
call-forward b2bua mailbox 20000
--- Define call forward mailbox number
call-forward b2bua noan 20000 timeout 15 --- Define call forward on no answer
allow watch
name Maria
--- Define directory name
label Maria
--- Define directory label
mwi
!
voice register dialplan 1
--- Create dial plan 1 for Cisco UCME SIP phones
type 7940-7960-others
--- Define a phone type for the SIP dial plan
pattern 1 777..
--- Define a dial pattern for Cisco UCME extensions
pattern 2 14...
--- Define a dial pattern for Avaya extensions
!
voice register pool 2
--- SIP phone settings
id mac 001E.4A34.D081
--- Enter SIP phone MAC address
AM; Reviewed:
SPOC 08/10/2009

35 of 48
ASM-CM-CUCME
type 7965
--- Define phone type
number 1 dn 2
--- Assign directory number 2 to phone line 1
dialplan 1
--- Assign dial plan to this phone pool
presence call-list
dtmf-relay rtp-nte
--- Configure dtmf-relay as rtp-nte (RFC 2833)
voice-class codec 1
--- Assign voice codec class 1 to the phone
speed-dial 1 14303 label "Avaya Digital - 14303"
speed-dial 2 14302 label "Avaya 9630 IP - 14302"
speed-dial 3 14304 label "Avaya One-X C - 14304"
blf-speed-dial 4 77701 label "Tony"
!
interface GigabitEthernet0/1
ip address 192.45.131.1 255.255.255.0
--- IP Address assigned to Cisco UCME gigabit interface
no ip redirects
no ip unreachables
no ip proxy-arp
ip route-cache flow
duplex auto
speed auto
media-type rj45
negotiation auto
no mop enabled
!
interface Service-Engine2/0
ip unnumbered GigabitEthernet0/1
service-module ip address 192.45.131.50 255.255.255.0
service-module ip default-gateway 192.45.131.1
no cdp enable
!
router eigrp 1
network 192.45.131.0
no auto-summary
!
ip default-gateway 192.45.131.2
ip forward-protocol nd
ip route 192.45.131.50 255.255.255.255 Service-Engine2/0
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
logging trap debugging
snmp-server community public RO
snmp-server location SIL
AM; Reviewed:
SPOC 08/10/2009

36 of 48
ASM-CM-CUCME
snmp-server contact x@xxxx.com

!
!--- Enable TFTP server & have these files available for 7965/7945 SIP & 7975 SCCP phones to download
tftp-server flash:term65.default.loads
tftp-server flash:SIP45.8-4-2S.loads
tftp-server flash:jar45sip.8-4-1-23.sbn
tftp-server flash:cvm45sip.8-4-1-23.sbn
tftp-server flash:apps45.8-4-1-23.sbn
tftp-server flash:cnu45.8-4-1-23.sbn
tftp-server flash:dsp45.8-4-1-23.sbn
tftp-server flash:SCCP75.8-4-1-23.loads
tftp-server flash:jar75sccp.8-4-1-23.sbn
tftp-server flash:cvm75sccp.8-4-1-23.sbn
tftp-server flash:apps75.8-4-1-23.sbn
tftp-server flash:cnu75.8-4-1-23.sbn
tftp-server flash:dsp75.8-4-1-23.sbn
!
control-plane
!
ccm-manager fax protocol cisco
!
mgcp fax t38 ecm
!
sccp local GigabitEthernet0/1 --- Set local interface that SCCP applications use to register with UCME
sccp ccm 192.45.131.1 identifier 1 version 7.0 --- Enable UCME to register SCCP applications
sccp
--- Enable SCCP and its related applications
!
sccp ccm group 1
--- Create UCME SCCP group
associate ccm 1 priority 1
--- Associate priority 1 to UCME
associate profile 1 register mtp001D45E95F20 --- Associates a DSP farm profile with UCME group
!
dspfarm profile 1 transcode
--- Define an application profile for DSP farm services.
codec g711ulaw
--- Specify G.711 u-law codec
codec g711alaw
--- Specify G.711 a-law codec
codec g729ar8
--- Specify G.729A codec
codec g729br8
--- Specify G.729B codec
maximum sessions 5
--- Specify maximum number of sessions
associate application SCCP
--- Associate SCCP with the DSP farm profile
!
! --- SIP Trunk Configuration
dial-peer voice 20000 voip
--- Create a VoIP dial-peer SIP Trunk to connect to voicemail
destination-pattern 2....
--- Set destination-pattern 2.
b2bua
--- Enable B2BUA on this dial-peer
AM; Reviewed:
SPOC 08/10/2009

37 of 48
ASM-CM-CUCME
session protocol sipv2

--- Set Session Protocol SIP Version 2
session target ipv4:192.45.131.50
--- Configure SIP session target for UCME
dtmf-relay sip-notify
--- Configure DTMF relay via SIP NOTIFY messages
codec g711ulaw
--- Assign G.711 u-law codec to dial-peer
no vad
!
--- Create a VoIP dial-peer SIP Trunk to connect to Avaya
description "Out to Avaya SM/CM"
--- Configure Description
destination-pattern 143..
--- Configure destination-pattern 143..
voice-class codec 1
--- Assign voice codec class 1 to the dial-peer
session target ipv4:10.1.2.170
--- Configure Avaya Aura Session Manager as session target
session transport tcp
--- Configure SIP session transport to TCP
dtmf-relay rtp-nte
!
--- Create an incoming VoIP dial-peer SIP Trunk
description "Incoming dial-peer"
--- Configure Description
voice-class codec 1
--- Assign voice codec class 1 to the dial-peer
--- Configure SIP session transport to TCP
session transport tcp
incoming called-number 777..
--- Configure incoming called 777..
dtmf-relay rtp-nte
!
presence
presence call-list
max-subscription 120
!
gateway
timer receive-rtp 1200
!
sip-ua
presence enable
!
telephony-service
--- SCCP global telephony-service settings for SCCP phones
sdspfarm units 5
--- Configure maximum # of DSP farms allowed to register
sdspfarm transcode sessions 8
--- Configure maximum # of G.729 transcoder sessions
sdspfarm tag 1 mtp001D45E95F20
--- Permits a DSP farm to register to UCME & associate it with a
SCCP phone MAC address
max-ephones 24
--- Set maximum number of phones that can register to UCME
max-dn 72
--- Set maximum number of directory numbers
ip source-address 192.45.131.1 port 3000 --- Set IP address and port # for UCME phone registration
system message SIL UCME
--- Configure a message for display on SCCP phones
load 7975 SCCP75.8-4-2S
--- Associate a 7975 SCCP phone type with a firmware file
time-zone 12
--- Configure time zone
voicemail 20000
--- Define voicemail access number
max-conferences 12 gain -6
--- Set maximum number of simultaneous 3-party conferences
AM; Reviewed:
SPOC 08/10/2009

38 of 48
ASM-CM-CUCME
call-forward pattern .....

--- Configure call forward pattern
moh music-on-hold.au
--- Configure Music-On-Hold (MOH) file
web admin system name interop secret 5 $1$Vtit$9esw1diEw.JuzfAgcLnd71
web admin customer name DE password bear
dn-webedit
time-webedit
transfer-system full-consult
--- Configure transfers using consultation, if available
transfer-pattern .T
--- Configure transfer pattern
secondary-dialtone 9
--- Configure secondary dial tone for outside line
create cnf-files
--- Create XML configuration files required for SCCP Phones
!
ephone-dn 11 dual-line
--- SCCP phone directory number settings
number 77701
--- Define directory number (extension)
label Tony
--- Define directory label
name Tony
--- Define directory name
allow watch
call-forward busy 20000
--- Define call forward busy
call-forward noan 20000 timeout 10
--- Define call forward on no answer
!
ephone 11
--- SCCP phone settings
mac-address 001D.45E9.5F20
--- Enter SCCP phone MAC address
username "tony" password 1234
--- Set username and password
presence call-list
blf-speed-dial 1 77710 label "Fred"
blf-speed-dial 2 77702 label "Maria"
speed-dial 1 14303 label "Avaya Digital - 14303"
speed-dial 2 14302 label "Avaya 9630 IP - 14302"
speed-dial 3 14304 label "Avaya One-X C - 14304"
type 7975
--- Define phone type
mwi-line 1
keep-conference endcall
--- Configure conference initiator to exit & leave other parties connected
button 1:11
--- Assign directory number 11 to button 1
pin 1234
--- Set username and password
!
banner login ^CAuthorized access only!
Disconnect IMMEDIATELY if you are not an authorized user!^C
!
line con 0
login local
transport output telnet
line aux 0
!
line con 0
login local
AM; Reviewed:
SPOC 08/10/2009

39 of 48
ASM-CM-CUCME
line aux 0
login local
line 130
no activation-character
no exec
transport preferred none
transport input all
transport output all
line vty 0 4
privilege level 15
login local
transport input all
line vty 5 15
privilege level 15
login local
!
scheduler allocate 20000 1000
end
After the configuration steps are complete, use the following commands to reset all SIP and
SCCP telephones to force them to load the configuration file.
configure t
voice register global
reset
exit
telephony-service
reset all
AM; Reviewed:
SPOC 08/10/2009

40 of 48
ASM-CM-CUCME
6 Verification Steps
This section provides the tests that can be performed on Avaya Aura Communication
Manager, Avaya Aura Session Manager, and Cisco UCME to verify proper their proper
configuration.
6.1
Verify Avaya Aura Communication Manager
Verify the status of the SIP trunk group by using the status trunk n command, where n is the
trunk group number administered in Section 3.5. Verify that all trunks are in the inservice/idle state as shown below.
status trunk 25
TRUNK GROUP STATUS
Member
Port
Service State
Mtce Connected Ports

Busy
0025/001
0025/002
0025/003
0025/004
0025/005
0025/006
0025/007
0025/008
0025/009
0025/010
T00226
T00227
T00228
T00229
T00230
T00231
T00232
T00233
T00234
T00235
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
no
no
no
no
no
no
no
no
no
no
Verify the status of the SIP signaling groups by using the status signaling-group n command,
where n is the signaling group number administered in Section 3.5. Verify the signaling group
is in-service as indicated in the Group State field shown below.
status signaling-group 25
STATUS SIGNALING GROUP
Group ID:
Group Type:
Signaling Type:
Group State:
AM; Reviewed:
SPOC 08/10/2009
25
sip
facility associated signaling
in-service
Active NCA-TSC Count: 0

Active CA-TSC Count: 0

41 of 48
ASM-CM-CUCME
Make a call between the Avaya 9600 Series IP Telephone and the Cisco 7965 SIP Telephone.
Verify the status of connected SIP trunks by using the status trunk x/y, where x is the
number of the SIP trunk group from Section 3.5.2 to reach Avaya Aura Session Manager, and
y is the member number of a connected trunk. Verify on Page 1 that the Service State is inservice/active. On Page 2, verify that the IP addresses of the S8300 Server and Avaya Aura
Session Manager are shown in the Signaling section. In addition, the Audio section shows the
IP addresses of the Avaya H.323 Telephone and Cisco UCME. The Audio Connection Type
displays ip-direct, indicating direct media between the two endpoints.
status trunk 25/1
Page
1 of
TRUNK STATUS
Service State: in-service/active
Maintenance Busy? no
Trunk Group/Member: 0025/001

Port: T00226
Signaling Group ID: 25
IGAR Connection? no
Connected Ports: S00504
status trunk 25/1
Page
2 of
CALL CONTROL SIGNALING

Near-end Signaling Loc: 01A0317
Signaling
IP Address
Near-end: 172.28.43.5
Far-end: 10.1.2.170
H.245 Near:
H.245 Far:
H.245 Signaling Loc:
Port
: 5061
: 5061
H.245 Tunneled in Q.931? no
Audio Connection Type: ip-direct

Near-end Audio Loc:
Audio
IP Address
Near-end: 172.28.43.103
Far-end: 192.45.131.1
Authentication Type: None

Codec Type: G.711MU
Port
: 6646
: 5200
Video Near:
Video Far:
Video Port:
Video Near-end Codec:
Video Far-end Codec:
AM; Reviewed:
SPOC 08/10/2009

42 of 48
ASM-CM-CUCME
6.2
Verify Avaya Aura Session Manager
Expand the Session Manager menu on the left and click SIP Monitoring. Verify as shown
below that none of the links to the defined SIP entities is down, indicating that they are all
reachable for all routing.
Under All Monitored SIP entities, select the appropriate SIP entities and verify that the
connection status is Up, as shown below for the Cisco UCME.
AM; Reviewed:
SPOC 08/10/2009

43 of 48
ASM-CM-CUCME
6.3
Verify Unified Communications Manager Express
The following commands can be used to troubleshoot calls over SIP trunks:
Show commands:
show ephone registered - verifies ephone registration.
show voice register all displays all SIP configuration and register information.
show call active voice brief - displays active call information for voice calls.
show voip rtp connection - displays RTP named-event packet information (e.g. callerID number, IP Address, and ports).
show sip-ua call - displays active call SIP user agent information.
Debug commands:
debug ccsip message - displays all SIP messages.
debug ccsip calls - displays SIP call trace information.
debug sccp message - displays the sequence of the SCCP messages.
debug voip rtp session named events - enables debugging for RTP named events
packets.
6.4
Verification Scenarios
Verification scenarios for the configuration described in these Application Notes included:
Basic calls between various telephones on Avaya Aura Communication Manager and
Cisco Unified Communications Manager Express can be made in both directions using
G.711MU, G.729, and G.729AB. For G.729 interoperability, the IP codec set on Avaya
Aura Communication Manager must include a version of the G.729 that Cisco UCME
supports.
Proper display of the calling and called party name and number information was verified for
all telephones with the basic call scenario.
Supplementary calling features were verified. The feature scenarios involved additional
endpoints on the respective systems, such as performing an unattended transfer of the SIP
trunk call to a local endpoint on the same site, and then repeating the scenario to transfer the
AM; Reviewed:
SPOC 08/10/2009

44 of 48
ASM-CM-CUCME
SIP trunk call to a remote endpoint on the other site. The supplementary calling features
verified are shown below. Note that calling/called party name and number display may
not be consistent in some cases.
o
o
o
o
o
o
Unattended transfer
Attended transfer
Hold/Unhold
Consultation hold
Call forwarding
Conference
AM; Reviewed:
SPOC 08/10/2009

45 of 48
ASM-CM-CUCME
7 Conclusion
As illustrated in these Application Notes, Avaya Aura Communication Manager can
interoperate with Cisco Unified Communications Manager Express using SIP trunks via Avaya
Aura Session Manager. The following is a list of interoperability items to note:
For G.729 interoperability, make sure both G.729 and G729AB are part of the audio codec
selection in Avaya Aura Communication Manager.
Audio shuffling between the Avaya H.323 IP telephones and Cisco UCME is supported.
Audio shuffling to the Cisco endpoints was not achieved during testing.
Calling and Called Party Name and Number displays may not be consistent in some cases for
calls involving transfers, conferences, and call forwarding.
Privacy calls between Avaya Telephones and Cisco SIP Telephones did not work as
expected:
The Called Party Name and Number were not displayed on the Avaya Telephones when
privacy was invoked on calls made from Avaya Telephones to Cisco SIP Telephones.
The Calling Party Name and Number were not blocked on calls from Cisco SIP endpoints
to Avaya endpoints with privacy invoked.
8 Additional References
This section references the product documentation relevant to these Application Notes.
Avaya Aura Session Manager:
[1]
Avaya Aura Session Manager Overview, Doc ID 03-603323, available at
http://support.avaya.com.
[2]
Installing and Administering Avaya Aura Session Manager, Doc ID 03-603324,
available at http://support.avaya.com.
[3]
Maintaining and Troubleshooting Avaya Aura Session Manager, Doc ID 03-603325,
Avaya Aura Communication Manager:
[4]
SIP Support in Avaya Aura Communication Manager Running on Avaya S8xxx
Servers, Doc ID 555-245-206, May, 2009, available at http://support.avaya.com.
[5]
Administering Avaya Aura Communication Manager, Doc ID 03-300509, May 2009,
AM; Reviewed:
SPOC 08/10/2009

46 of 48
ASM-CM-CUCME
Product documentation for Cisco Systems products may be found at http://www.cisco.com

[1]
[2]
[3]
[4]
[5]
Cisco Unified Communications Manager Express System Administrator Guide, February

27, 2009, Part Number: OL-10663-02
Cisco Unified Communications Manager Express Command Reference Guide, February
27, 2009, Part Number: OL-10894-01
Cisco CallManager Express (CME) SIP Trunking Configuration example, November 16,
2007, Document ID: 91535
Cisco Unified CME Solution Reference Solution Design Guide, Release 7.0(1), Part
Number: OL-1062101-01
Cisco Unified Communications Manager Express: SIP Implementation Guide, November
9, 2007, Document ID: 99946
AM; Reviewed:
SPOC 08/10/2009

47 of 48
ASM-CM-CUCME

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 Solution &
Interoperability Test Lab at interoplabnotes@list.avaya.com
AM; Reviewed:
SPOC 08/10/2009

48 of 48
ASM-CM-CUCME

Asm CM Cucme

Transféré par

Informations du document

Titre original

Copyright

Formats disponibles

Partager ce document

Partager ou intégrer le document

Options de partage

Avez-vous trouvé ce document utile ?

Ce contenu est-il inapproprié ?

Droits d'auteur :

Formats disponibles

Asm CM Cucme

Transféré par

Droits d'auteur :

Formats disponibles

Avaya Solution & Interoperability Test Lab

Configuring SIP Trunks among Avaya Aura Session

Solution & Interoperability Test Lab Application Notes

Figure 1 Sample Configuration

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

2 Equipment and Software Validated

3 Configure Avaya Aura Communication Manager

Verify Avaya Aura Communication Manager license

Some administration screens have been abbreviated for clarity.

Solution & Interoperability Test Lab Application Notes

Verify Avaya Aura Communication Manager License

Configure System Parameters Features

Solution & Interoperability Test Lab Application Notes

Configure IP Node Names

Configure IP Codec Sets and Network Regions

Solution & Interoperability Test Lab Application Notes

Configure SIP Signaling Group and Trunk Group

3.5.1 SIP Signaling Group

DTMF over IP:

Group Type: sip

Far-end Node Name: ASM

Direct IP-IP Audio Connections?

Solution & Interoperability Test Lab Application Notes

3.5.2 SIP Trunk Group

Group Type: sip

Numbering Format: public

Solution & Interoperability Test Lab Application Notes

Configure Route Pattern

ITC BCIE Service/Feature PARM

No. Numbering LAR

Solution & Interoperability Test Lab Application Notes

Configure Location and Public Unknown Numbering

A descriptive name to denote the Main site.

Solution & Interoperability Test Lab Application Notes

Administer Uniform Dial Plan and AAR Analysis

Dialed prefix digits to match on, in this case 777.

Dialed prefix digits to match on, in this case 777.

change aar analysis 0

Configuration of Avaya Aura Communication Manager is complete. Use the save

Solution & Interoperability Test Lab Application Notes

4 Configure Avaya Aura Session Manager

Configuration is accomplished by accessing the browser-based GUI of Avaya Aura System

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Specify SIP Domain

The authoritative domain name (e.g., avaya.com)

Solution & Interoperability Test Lab Application Notes

Under Location Pattern:

A pattern used to logically identify the location.

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

For the Cisco UCME adaptation, enter the following information.

Solution & Interoperability Test Lab Application Notes

Add SIP Entities

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Solution & Interoperability Test Lab Application Notes

Add Entity Links

Solution & Interoperability Test Lab Application Notes