Avaya Solution & Interoperability Test Lab
Application Notes for Configuring a SIP Trunk between Avaya Communication Manager and Cisco Unified Communications Manager Express - Issue 1.0
Abstract These Application Notes present a sample configuration of a SIP Trunk between an Avaya Communication Manager and Cisco Unified Communications Manager Express. The configuration described is applicable to other Avaya Servers and Media Gateways that run Avaya Communication Manager.
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1. Introduction These Application Notes present a sample configuration for a network comprised of an Avaya S8710 Server IP Connect configuration and a Cisco Unified Communications Manager Express (CME). The focus is on the configuration of the SIP Signaling Group and IP Trunk Group on Avaya Communication Manager and the corresponding configuration of the Cisco Unified Communications Manager Express. Using the configuration described herein, Cisco IP Telephones controlled by the Cisco Unified Communications Manager Express can call (and be called by) Avaya IP Telephones and endpoints controlled by Avaya Communication Manager. Figure 1 shows the network setup used for these configurations. Avaya S8710 Server
Cisco Catalyst 4503 Switch Cisco Call Manager Express Cisco 3825 Router
Avaya G650 Media Gateway
N C-LA IPSI
Si
RO MEDP
Cisco Catalyst Express 500 Switch
Cisco 2811 Router 1 2
3 4
5 6
7 8
9 10
11 12
13 14
15 16
17 18
19 20
21 22
CatalystExpress500GSeries
23 24
1X
11X 13X
23X
2X
12X
24X
SYSTEM ALERT
1 16X
14X
2
1X 6
SETUP
CISCO IP PHONE 7960
1
2 ABC
Avaya 2420 DigitalTelephone X50002
Avaya 4621 IP Telephone X50000
Avaya 9630 IP Telephone X50008
3 DEF
4
5
GHI
JKL
MNO
7 PQRS
8 TUV
9 WXYZ
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CISCO IP PHONE 7960
messages i
directories
services
settings
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Cisco 7961G IP Telephone X77701 X77702
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DEF
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GHI
JKL
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8 TUV
9 WXYZ
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messages i
directories
services
settings
MNO
OPER
Cisco 7961G IP Telephone x77710
Figure 1: Avaya-Cisco SIP Interoperability Configuration
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Table 1 shows the device interfaces and IP network assignment. Network Component Avaya S8710 Server1 Avaya S8710 Server2 G650 Gateway C-LAN MEDPRO Avaya 9630 one-X IP Telephone Avaya 4621 IP Telephone Cisco Unified Communications Manager Express Cisco 2811 Router Cisco 7961G IP Telephone 1 Cisco 7961G IP Telephone 2
IP Address 192.168.1.2 192.168.1.3
Gateway 192.168.1.1
Network Mask 255.255.255.0
192.168.1.10 192.168.1.11 192.168.1.201 192.168.1.202 192.45.130.3
192.168.1.1 192.168.1.1 192.168.1.1 192.168.1.1 192.45.130.1
255.255.255.0
192.45.130.1 DHCP
255.255.255.0 255.255.255.0
Table 1: Device Interfaces and Network IP Address Assignment
2. Hardware and Software Used for Verification Table 2 lists the equipment and software used for the verification. Equipment Avaya S8710 Server running Avaya Communication Manager Avaya G650 Media Gateway with • C-LAN • MEDPRO Avaya 9630 one-X IP Telephone Avaya 4621 IP Telephone Cisco 2811 Router Cisco Catalyst Express 500 Switch Cisco Unified Communications Manager Express Cisco 7961G IP Telephone
Software R014x.00.1.731.2
HW01 FW024 HW20 FW117 R1.5 R2.8 IOS 12.4(15)T1 12.2.25-SEG4 IOS 12.4(11)XW2 SIP 8.2.1
Table 2: Hardware and Software Used for Verification
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3. Avaya Communication Manager Configuration This section presents configuration steps for Avaya Communication Manager. It is assumed that Avaya Communication Manager has been installed and the login and password credentials are available to the reader. All the commands discussed in this section are executed on Avaya Communication Manager using the System Access Terminal (SAT).
3.1. Add Node Name and Map IP Address The following configuration displays a subset of the “change node-names ip” screen that maps logical names to IP address. These node names are presented because they will appear in other screens, such as defining the SIP signaling group to the Cisco Unified Communication Manager. change node-names ip
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IP NODE NAMES Name CCM-Express c-lan Medpro
IP Address 192.45.131.1 192.168.1.10 192.168.1.11
3.2. Configure C-LAN and MEDPRO Uses the command add ip-interface to add and configure the C-LAN and the MEDPRO of the Avaya G650 Media Gateway. The following two screens display the configurations of the CLAN (01A02) and the MEDPRO (01A03). Note that the C-LAN and MEDPRO are assigned to Network Region 1. display ip-interface 1a02 IP INTERFACES
Type: Slot: Code/Suffix: Node Name: IP Address: Subnet Mask: Gateway Address: Enable Ethernet Port? Network Region: VLAN:
C-LAN 01A02 TN799 D c-lan 192.168.1 .10 255.255.255.0 192.168.1 .1 y 1 192
Link: 1 Allow H.323 Endpoints? y Allow H.248 Gateways? y Gatekeeper Priority: 5
Target socket load and Warning level: 400 Receive Buffer TCP Window Size: 8320 ETHERNET OPTIONS Auto? y
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display ip-interface 1a03 IP INTERFACES
Type: Slot: Code/Suffix: Node Name: IP Address: Subnet Mask: Gateway Address: Enable Ethernet Port? Network Region: VLAN:
MEDPRO 01A03 TN2302 Medpro 192 .168 .1 .11 255.255.255.0 192 .168 .1 .1 y 1 192 ETHERNET OPTIONS
Auto? y
3.3. Configure IP Codec In these Application Notes, two IP network regions are configured. Region 1 is used for Avaya location and region 7 is used for Cisco Unified Communications Manager location. Codec G.711ulaw is used within each region 1 and G.729A is used between these two regions. The following screens display the configuration for IP codec set 1 and 7. display ip-codec-set 1
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IP Codec Set Codec Set: 1 Audio Silence Codec Suppression 1: G.711MU n 2: G.729A n 3: Media Encryption 1: none 2:
Frames Per Pkt 2 2
Packet Size(ms) 20 20
display ip-codec-set 7
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IP Codec Set Codec Set: 7 Audio Silence Codec Suppression 1: G.729A n 2: G.711MU n 3: Media Encryption 1: none 2:
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Frames Per Pkt 2 2
Packet Size(ms) 20 20
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3.4. Configure the Network Regions The following illustrates the configuration for network region 1. The intent of illustrating the network region is to show that Codec Set 1 is configured and the “Direct IP-IP Audio Connections” field for Intra-region is set to “yes”. The Inter-region IP-IP Direct Audio is set to “yes” to ensure that the media path will terminate on Avaya IP Telephones. change ip-network-region 1
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IP NETWORK REGION Region: 1 Location: Authoritative Domain: avaya.com Name: 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? y UDP Port Max: 2999 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 26 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: 3 Audio 802.1p Priority: 5 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
Note that on page 3, codec 1 is used in region 1 and codec 7 is used between the region 1 and region 7. change ip-network-region 1
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Inter Network Region Connection Management src rgn 1 . 1
dst codec direct WAN-BW-limits Video rgn set WAN Units Total Norm Prio Shr Intervening-regions 1 1 . 7 7 y NoLimit
Dyn CAC IGAR
n
The following screen shows the configuration for network region 7. The intent of illustrating the network region is to show that Codec Set 7 is configured and the “Direct IP-IP Audio Connections” field for Intra-region is set to “yes”. The Inter-region IP-IP Direct Audio is set to “yes” to ensure that the media path will terminate on Avaya Telephones.
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change ip-network-region 7
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IP NETWORK REGION Region: 7 Location: Authoritative Domain: Name: CCM-Express MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 7 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 3029 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
Also, on page 3 of 19, the codec 7 is selected for calls between regions 1 and 7. change ip-network-region 7
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Inter Network Region Connection Management src dst codec direct rgn rgn 7 1 . . 7 7
set 7
WAN y
WAN-BW-limits
Units NoLimit
Total Norm
Video
Dyn
Prio Shr Intervening-regions
CAC IGAR n
1
3.5. Configure IP Network Map Use “change ip-network-map” command to put all devices that are on 192.168.1.0 network (Avaya site) into the region 1 as shown below. change ip-network-map
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IP ADDRESS MAPPING
From IP Address 192 .168 .1 .1
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Subnet (To IP Address or Mask) 192 .168 .1 .254
Region 1
VLAN n
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Emergency Location Extension
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3.6. Configure SIP Signaling Group This section focuses on the parameter settings recommended for the SIP signaling group and IP trunk group used to connect with the Cisco Unified Communications Manager Express. Signaling group 4 is created to establish a SIP signaling link between a C-LAN in the G650 Media Gateway and the Cisco Unified Communications Manager Express. The signaling group number is not relevant; use any available signaling group number. Use the command “add signaling-group 4” to add the signaling group. This signaling group uses the C-LAN whose node-name is “c-lan” as the near end, and the Cisco Unified Communications Manager Express node-name “CCM-Express” as the far-end note name. Both node-names were defined in Section 3.1. Use TCP as “Transport Method” and keep the default listen port 1720 for both “Near-end Listen Port “ and “Far-end Listen Port”. Note: Do not use TLS for “Transport Method” since it’s not supported by CME. The “Direct IP-IP Audio Connections?” field is set to “yes” to allow the media path for a call to be “direct” from Avaya Telephones to Cisco IP Telephone. Select “rtp-payload” as DTMF over IP. The far-end network region field can optionally be populated with a network region number to associate with the Cisco Unified Communications Manager Express. For the signaling group shown here, the far-end network region is set to 7 so that the calls between region 1 and region 7 will use codec set 7 (G.729A) as configured. change signaling-group 4
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SIGNALING GROUP Group Number: 4
Group Type: sip Transport Method: tcp Near-end Node Name: c-lan Far-end Node Name: CCM-Express Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 7 Far-end Domain: avaya.com Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload
Direct IP-IP Audio Connections? y IP Audio Hairpinning? y
Enable Layer 3 Test? n Session Establishment Timer(min): 3
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3.7. Configure IP Trunk Group Use the command “add trunk-group 4” to create a SIP trunk group.. Data has been entered in the fields shown in bold. Note that “Number of Members” is set to 10, which can carry 10 VoIP calls simultaneously on the IP trunk group. add trunk-group 4
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TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type:
4 SIP-CME two-way n 0 tie
Group Type: sip CDR Reports: y COR: 1 TN: 1 TAC: 104 Outgoing Display? n Night Service: Auth Code? n Signaling Group: 4 Number of Members: 10
On page 3, set private for “Numbering Format” as shown below. add trunk-group 4
Page
TRUNK FEATURES ACA Assignment? n
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Measured: none Maintenance Tests? y
Numbering Format: private UUI Treatment: service-provider
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y
add trunk-group 4
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PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Telephone Event Payload Type: 127
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On Page 5 of the trunk configuration, Avaya Communication Manager adds the 10 trunk group members autumnally based on the member configured on page 1. add trunk-group 4
Page 5 of 21 TRUNK GROUP Administered Members (min/max): 1/10 Total Administered Members: 10
GROUP MEMBER ASSIGNMENTS
1: 2: 3: 4: 5: 6: 7: 8: 9: 10: 11:
Port T00044 T00045 T00046 T00047 T00048 T00049 T00050 T00051 T00052 T00053
Name SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME SIP-CME
3.8. Configure Uniform Dial-Plan and Route Pattern Use command “change uniform-dialplan 77” to set Avaya Communication Manager (ACM) to route call using AAR table for extensions starting digits 77. change uniform-dialplan 77
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UNIFORM DIAL PLAN TABLE Percent Full: 0 Matching Pattern 77
Len Del 5 0
Insert Digits
Node Net Conv Num aar n
Route pattern 77 is created on the Communication Manager to route calls to Cisco Unified Communications Manager Express. With the configuration displayed below, Avaya Communication Manager will route calls with destination 77xxx using trunk group 4 configured in the previous sections. change route-pattern 77 Page 1 of 3 Pattern Number: 10 Pattern Name: To Cisco CME SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 4 0 n user 2: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM 0 1 2 M 4 W Request 1: y y y y y n 2: y y y y y n
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n n
No. Numbering LAR Dgts Format Subaddress
rest rest
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none none
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Use command “change aar analysis” to set aar table to use route pattern 77 for dialed string starting with 77. change aar analysis 77
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AAR DIGIT ANALYSIS TABLE Percent Full: Dialed String 77
Total Min Max 5 5
Route Pattern 77
Call Type aar
Node Num
2
ANI Reqd n
In order for Avaya Communication Manager to send the calling number to Cisco CME via this IP trunk, use the command “change public-unknown-number” to enable this feature as shown below. change public-unknown-numbering 5
Ext Ext Len Code 5
5
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NUMBERING - PUBLIC/UNKNOWN FORMAT Total Trk CPN CPN Grp(s) Prefix Len Total Administered: 2 4 5 Maximum Entries: 9999
The command “save translation” must be entered to save the configuration changes.
4. Cisco Unified Communications Manager Express Configuration Cisco Unified Communications Manager Express is a call-processing application in Cisco IOS software that enables Cisco routers to deliver key-system or hybrid 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 Unified Communications Manager Express configuration for SIP trunking to Avaya Communication Manager. A VoIP dial peer will be configured in the Cisco Unified Communications Manager Express to connect to the IP Address of the C-LAN in the Avaya G650 Media Gateway.
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With the Cisco IOS 12.4(11)XW2 used in this configuration, Cisco 7961G SIP IP telephones require the following firmware to work with the CME. • • • • • • • •
SIP41.8-2-1S.loads term41.default.loads term61.default.loads apps41.8-0-2-55.sbn cnu41.8-2-0-55.sbn cvm41sip.8-2-0-55.sbn dsp41.8-2-0-55.sbn jar41sip.8-0-2-25.sbn
This section only presents the VoIP related configuration on the Cisco 3825 router. hostname Care3825 ! boot-start-marker boot system flash c3825-ipvoicek9-mz.124-11.XW2.bin boot-end-marker ! 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 cme import all network 192.45.131.0 255.255.255.0 default-router 192.45.131.1 option 150 ip 192.45.131.2 --- Use option 150 to set CME as TFTP server ! ! no ip bootp server no ip domain lookup ip domain name careslab.local ip name-server 135.8.31.182 ! voice-card 0 no dspfarm ! interface GigabitEthernet0/1 ip address 192.45.131.2 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip route-cache flow SZ; Reviewed: SPOC 8/13/2008
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duplex auto speed auto media-type rj45 negotiation auto no mop enabled ! tftp-server flash:SIP41.8-2-1S.loads tftp-server flash:jar41sip.8-2-0-55.sbn tftp-server flash:dsp41.8-2-0-55.sbn tftp-server flash:cvm41sip.8-2-0-55.sbn tftp-server flash:cnu41.8-2-0-55.sbn tftp-server flash:apps41.8-2-0-55.sbn tftp-server flash:term61.default.loads tftp-server flash:term41.default.loads ! dial-peer voice 50000 voip destination-pattern 5.... session protocol sipv2 session target ipv4:192.168.1.10 session transport tcp dtmf-relay rtp-nte !
--- enable TFTP server and have these files available for 7961G/7941 telephone to download
--- Create a VoIP dial-peer --- Set destination-pattern 5…. --- Set Session Protocol SIP Version 2 --- Enable Far-end SIP signaling endpoint on c-lan (192.168.1.10 --- Use TCP as transport protocol --- set dtmf RTP Named Telephone Event RFC 2833
--- Enable voice register global configuration mode to set parameters for all supported SIP phones in Cisco --- Unifed CME voice register global mode cme source-address 192.45.131.1 port 5060
--- enable mode for provisioning SIP phones on CME --- enable CME router to receive SIP phone messages
through the specified IP address and port. max-dn 50 --- define max dn number supported on CME max-pool 20 --- limited the number of SIP phones supported by CME. load 7961 sip41.8-2-1S --- associate a phone type with a phone firmware file dialplan-pattern 1 50... extension-length 5 --- define dialplan-pattern for SIP phone call
dialplan-pattern 2 777.. extension-length 5 external-ring bellcore-dr3 create profile --- create a profile on CME !
--- enter voice register dn configuration mode to define a directory number for a SIP phone ----voice register dn 1 --- configure dn 1 number 77701 --- assign extension 77701 to this dn SZ; Reviewed: SPOC 8/13/2008
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label John --- assign phone lable “John” as user name mwi --- enable mwi for this dn. ! voice register dn 2 number 77702 mwi ! voice register dn 3 number 77710 label Fred mwi ! --- enter voice register dialplan configuration mode to define a dial plan for SIP phones --voice register dialplan 1 --- create a dialplan 1 type 7940-7960-others --- define a phone type for the SIP dial plan. --- Note: 7940-7960-others are defined for these Cisco IP phones: 7911, --- 7940, 7940G, 7941, 7941GE, 7960, 7960G, 7961, 7961GE, 7970, or 7971.
pattern 1 50... pattern 2 777..
--- define a dial pattern 50… for SIP dial plan for Avaya phones. --- define a dial pattern 777.. for SIP phones registered on CME.
! --- enter voice register pool configuration mode to set phone-specific parameters for a SIP phone --voice register pool 1 --- create a voice register pool 1 id mac 0019.5628.43F9 --- enter SIP phone MAC address in this pool. type 7961GE --- define phone type. number 1 dn 1 --- assign directory number 1 to phone line 1. number 2 dn 2 --- assign directory number 2 to phone line 2. dialplan 1 ---- use dialplan 1 for this pool dtmf-relay rtp-nte --- configure dtmf-relay as rtp-nte ! voice register pool 2 id mac 0015.2B34.0B1C --- enter SIP phone MAC address in this pool. type 7961GE number 1 dn 3 --- assign directory number 3 to phone line 1. dialplan 1 dtmf-relay rtp-nte ip route 0.0.0.0 0.0.0.0 192.45.131.1
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After the configuration steps are complete, use the following command to reset all SIP telephones to force them to load the configuration file. configure t voice register global reset
5. Verification Steps The following steps can be used to verify the configuration described in these Application Notes. •
Make a phone call from the Avaya 9630 IP phone (50008) to the Cisco 7961 IP phone (77701), and verify the voice quality is good and the SIP trunk is used to carry this call. From SAT, the command “status station 50008” displays the call signaling and audio information.
status station 50008 Administered Type: Connected Type: Extension: Port: Call Parked? Ring Cut Off Act? Active Coverage Option:
Page 1 of 7 GENERAL STATUS Service State: in-service/off-hook TCP Signal Status: connected
4620 9640 50008 S00026 no no 1
EC500 Status: N/A Message Waiting: Connected Ports: T00044
Parameter Download: complete SAC Activated? no
Off-PBX Service State: N/A
Limit Incoming Calls? no User Cntrl Restr: none Group Cntrl Restr: none
HOSPITALITY STATUS Awaken at: User DND: not activated Group DND: not activated Room Status: non-guest room
status station 50008
Page 3 of 7 CALL CONTROL SIGNALING Port: S00026 Switch-End IP Signaling Loc: 01A0217 H.245 Port: IP Address Port Node Name Rgn Switch-End: 192.168. 1. 10 61440 c-lan 1 Set End: 192.168. 1.201 1720 1 H.245 Near: H.245 Set: status station 50008
Page 4 of 7 AUDIO CHANNEL Port: S00026 G.729A Switch-End Audio Location: IP Address Port Node Name Rgn Other-End: 192. 45.131. 1 18214 CCM-Express 7 Set-End: 192.168. 1.201 2868 1 Audio Connection Type: ip-direct
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•
When the call is up, use command “status trunk 4” to verify that the call is carried by trunk 4. The channel is in active status.
status trunk 4 TRUNK GROUP STATUS Member
Port
Service State
Mtce Connected Ports Busy
0004/001 0004/002 0004/003 0004/004 0004/005 0004/006 0004/007 0004/008 0004/009 0004/010
T00044 T00045 T00046 T00047 T00048 T00049 T00050 T00051 T00052 T00053
in-service/active 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
•
S00026
Use command “status trunk 4/1” to verify the signaling path is between C-LAN and the Cisco CME and the audio path is between Avaya 9630 IP Telephone (192.168.1.201) and the Cisco CME (192.45.131.1). The codec used is G.729.
status trunk 4/1
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TRUNK STATUS Trunk Group/Member: 0004/001 Port: T00044 Signaling Group ID:
Service State: in-service/active Maintenance Busy? no
IGAR Connection? no Connected Ports: S00026 Port Signaling: 01A0217 G.729
Audio: Video: Video Codec:
Near-end IP Addr : Port 192.168. 1. 10 : 5060
Far-end IP Addr : Port 192. 45.131. 1 : 5060
192.168.
192. 45.131.
1.201
: 2868
1 : 18214
Authentication Type: None Audio Connection Type: ip-direct
status trunk 4/1
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SRC PORT TO DEST PORT TALKPATH src port: T00044 T00044:TX:192.45.131.1:18214/g729/20ms S00026:RX:192.168.1.201:2868/g729a/20ms
dst port: S00026
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•
Make a phone call from the Cisco 7961G (77710) IP phone to the Avaya digital phone (50002), and verify the voice quality is good. Transfer the call to Avaya IP 9630 telephone (50008) and verify that the transfer is successful.
•
Make a phone call from the Avaya IP 9630 telephone (extension 50008) to Cisco 7961G telephone (extension 77701). While the call is up, conference the Cisco telephone (extension 77710) from the Cisco telephone (extension 77701) and verify that all three parties are in conference.
•
Set up a meet-me conference on Avaya Communication Manager. Dial the meet-me conference number from the Cisco 7961G IP Telephone (x77710) and enter the password after the prompt. Verify that DTMF is working as the Cisco 7961G telephone can join the conference successfully.
6. Conclusion As illustrated in these Application Notes, Avaya Communication Manager can interoperate with the Cisco Unified Communications Manager Express using a SIP trunk to carry VoIP calls between Avaya and Cisco telephones. The calling party name and number can be displayed for calls in both directions.
7. Additional References [1] Cisco Unified Communications Manager Express System Administrator Guide http://cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter0918 6a00807d5548.html [2] Administrator Guide for Avaya Communication Manager, Document ID: 03-300509, Issue 3.1, February 2007 http://support.avaya.com/elmodocs2/comm_mgr/r4_0/pdfs/03_300509_3_1.pdf
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©2008 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 Solution & Interoperability Test Lab at
[email protected]
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