Cxc550 Sip PSTN

Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Direct SIP Trunking from Avaya Communication Manager to a Covergence CXC Session Manager Issue 1.0
Abstract
These Application Notes describe the configuration of direct SIP trunking from Avaya Communication Manager to a Covergence CXC Session Manager and a Cisco AS5400 Universal Gateway. The gateway provided ISDN PRI trunks to a telecommunications service provider network for PSTN interoperability. In this configuration, an Avaya SIP Enablement Services (SES) server is not used as part of the SIP trunking solution.
FS; Reviewed: SPOC 12/6/2007
Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved.
1 of 51 CXC550_SIP_PSTN
1. Introduction
These Application Notes describe the configuration of direct SIP trunking from Avaya Communication Manager to a Covergence CXC Session Manager and a Cisco AS5400 Universal Gateway. The gateway provided ISDN PRI trunks to a telecommunications service provider network for PSTN interoperability. In this configuration, an Avaya SIP Enablement Services (SES) server is not used as part of the SIP trunking solution. Direct SIP trunking uses the Covergence CXC Session Manager features to distribute SIP signaling for incoming calls to multiple Avaya Communication Manager C-LAN interfaces. This provides for additional capacity, load balancing, and survivability options. In addition, the Covergence CXC Session Manager performs conversion between the TCP transport for SIP messages used by the Avaya Communication Manager and the UDP transport commonly used by other communication elements and service provider networks. The configuration tested is shown in Figure 1. The Covergence CXC Session Manager functions as a SIP Back-To-Back User Agent (B2BUA), and acts as an intermediary to manage the SIP signaling and RTP media packets between Avaya Communication Manager and other SIP terminations, such as the Cisco AS5400 Universal Gateway or a SIP trunk directly to a telecommunication service provider. It often resides at the boundary of the enterprise customers IP network and serves as a security device to isolate the customers internal network from the public domain. Within these Application Notes, the networking was configured in a similar manner; the subnet used by Avaya Communication Manager was not directly connected to the subnet used by the Cisco AS5400 Universal Gateway and all SIP and RTP packets were routed through the Covergence CXC Session Manager, configured as a High Availability (HA) redundant server pair. The private side of the Covergence CXC Session Manager is the 10.1.1.0 network, while the public side is the 142.16.58.0 network. Although not used in the sample configuration, the Covergence CXC Session Manager is also capable of performing Network Address Translation (NAT), topology hiding, and encryption. In addition, RTP streams do not have to be routed through the Covergence CXC Session Manager. These Application Notes address the following capabilities: Incoming PSTN calls to Avaya H.323 IP and digital telephones. Outbound PSTN calls from Avaya H.323 IP and digital telephones. Trunk to trunk forwarding (tandem routing) of an inbound PSTN call to another PSTN telephone via Avaya Communication Manager. G.711mu-law and G.729A codecs. Direct IP-IP media between H.323 IP telephones and SIP trunks. Direct IP-IP media for SIP trunk to SIP trunk forwarded calls. Use of call center functionality (ACD splits, announcements, vectors, etc.). Load balancing of incoming calls across multiple C-LAN SIP trunks. Alternate routing upon failure of C-LAN SIP trunks for incoming and outgoing calls.
Figure 1 Direct SIP Trunking Test Configuration
2. Equipment and Software Validated

The following products and software were used for the configuration in Figure 1. Component Avaya S8500B Media Server Avaya G650 Media Gateway TN2312BP IP Server Interface (IPSI) TN799DP Control-LAN (C-LAN) TN2602AP IP Media Resource 320 (MEDPRO) TN2224CP Digital Line Avaya 4621SW IP (H.323) Telephones Avaya 6424D Digital Telephone Covergence CXC-550 Session Manager Cisco AS5400 Universal Gateway Version Avaya Communication Manager 4.0.1, Patch 14300 HW12 FW036 HW01 FW017 HW02 FW025 HW08 FW015 Release 2.8.3 3.2.12, Patch 29598 12.4(17)
Table 1 Equipment and Versions
3. Configure Avaya Communication Manager

Avaya Communication Manager was installed and configured for basic station to station calling and call center operation prior to beginning the configuration shown in these Application Notes. These basic configuration details are outside the scope of the SIP trunking application and not included here.
3.1. SIP Trunk Configuration

3.1.1. Verify System Capacity and Required Features
The Avaya Communication Manager license controls the customer options. Contact an authorized Avaya sales representative for assistance if insufficient capacity exists or a required feature is not enabled. Verify that there is sufficient remaining SIP trunk capacity available for the Cisco AS5400 Universal Gateway as well as any other SIP trunking applications in use. This is done by displaying Page 2 of the System-Parameters Customer-Options form. The number of SIP trunks available to assign to new or existing trunk groups is the difference between the Maximum Administered SIP Trunks and the USED value.
change system-parameters customer-options OPTIONAL FEATURES 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 Media Gateway VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: 100 100 0 0 0 0 0 0 100 0 10 0 128 128 0 Page 2 of 10
USED 10 5 0 0 0 0 0 0 50 0 1 0 2 0 0
(NOTE: You must logoff & login to effect the permission changes.)
Figure 2: System-Parameters Customer-Options Form Page 2
Verify that the Automatic Route Selection (ARS) feature is enabled on Page 3 of the SystemParameters Customer-Options form.
change system-parameters customer-options OPTIONAL FEATURES Abbreviated Dialing Enhanced List? Access Security Gateway (ASG)? Analog Trunk Incoming Call ID? A/D Grp/Sys List Dialing Start at 01? Answer Supervision by Call Classifier? ARS? ARS/AAR Partitioning? ARS/AAR Dialing without FAC? ASAI Link Core Capabilities? ASAI Link Plus Capabilities? Async. Transfer Mode (ATM) PNC? Async. Transfer Mode (ATM) Trunking? ATM WAN Spare Processor? ATMS? Attendant Vectoring? n n n n n y y n n n n n n n n Page 3 of 10
Audible Message Waiting? Authorization Codes? CAS Branch? CAS Main? Change COR by FAC? Computer Telephony Adjunct Links? Cvg Of Calls Redirected Off-net? DCS (Basic)? DCS Call Coverage? DCS with Rerouting?
n n n n n n n n n n
Digital Loss Plan Modification? n DS1 MSP? n DS1 Echo Cancellation? n
(NOTE: You must logoff & login to effect the permission changes.)
Figure 3: System-Parameters Customer-Options Form Page 3
3.1.2. Determine Node Names

Use the change node-names ip command to view (or assign) the node names to be used in the SIP trunk configuration. Covergence and 10.1.1.100 are the Name and IP Address of the virtual privateside interface of the Covergence CXC Session Manager, where Avaya Communication Manager SIP trunk messages are sent. clan_01a02 and 192.168.99.52 are the Name and IP Address of the TN799DP C-LAN interface used for the first SIP signaling group with the Covergence CXC Session Manager. clan_01a03 and 192.168.99.53 are the Name and IP Address of the TN799DP C-LAN interface used for the second SIP signaling group with the Covergence CXC Session Manager.
Page IP NODE NAMES Name Covergence clan_01a02 clan_01a03 default medpro_01a04 procr val_01a07 IP Address 10.1.1.100 192.168.99.52 192.168.99.53 0.0.0.0 192.168.99.54 192.168.99.50 192.168.99.57 1 of 2
change node-names ip
Figure 4: IP Node Names
3.1.3. Define IP Codec Sets

G.729A and G.711mu-law codecs (in that priority) are used for voice calls via the SIP trunks to the Cisco AS5400 Universal Gateway. This is IP codec set 1. Using the change ip-codec-set 1 command, enter G.729A and G.711MU as the first and second Audio Codec values on the form. Retain the defaults for the remaining fields.
change ip-codec-set 1 IP Codec Set Codec Set: 1 Audio Codec 1: G.729A 2: G.711MU 3: Silence Suppression n n Frames Per Pkt 2 2 Packet Size(ms) 20 20 Page 1 of 2
Figure 5: IP Codec Set 1
3.1.4. Configure the C-LAN IP Network Region Assignment

In these Application Notes, two C-LANs are assumed to been previously installed in board slots 1a02 and 1a03 as part of the initial Avaya Communication Manager basic installation (using the procedures as described in [2]) and assigned the Node Names shown in Figure 4. The configuration in this section will assign them to Network Region 1, used for the Direct SIP Trunking application. Use the change ip-interface ucss command (where u is the cabinet, c is carrier, and ss is the slot of the respective C-LAN), to assign each C-LAN to Network Region 1. Note: In order to change an existing IP interface, disable the interface by setting the Enable Ethernet Port field n, submit the change, and then use the change ip-interface ucss command again. The Enable Ethernet Port must then be re-enabled with y when the Network Region value is set.
change ip-interface 1a02 IP INTERFACES Page 1 of 1
Type: Slot: Code/Suffix: Node Name: IP Address: Subnet Mask: Gateway Address: Enable Ethernet Port? Network Region: VLAN:
C-LAN 01A02 TN799 D clan_01a02 192.168.99.52 255.255.255.0 . . . y 1 n
Link: 12 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? n Speed: 100Mbps Duplex: Full
Figure 6: IP Interface of C-LAN 1a02 used for SIP Signaling Group 11

FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 6 of 51 CXC550_SIP_PSTN
Use the command shown below for C-LAN 1a03, used for SIP trunk signaling group 12.
change ip-interface 01a03 IP INTERFACES Type: Slot: Code/Suffix: Node Name: IP Address: Subnet Mask: Gateway Address: Enable Ethernet Port? Network Region: VLAN: C-LAN 01A03 TN799 D clan_01a03 192.168.99.53 255.255.255.0 . . . y 1 n Page 1 of 1
Link: 13 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? n Speed: 100Mbps Duplex: Full
Figure 7: IP Interface of C-LAN 1a03 used for SIP Signaling Group 12
3.1.5. Define IP Network Region

IP network regions set IP network properties for SIP trunk groups and other IP elements (such as IP telephones, media processor cards, etc.) assigned to the region. IP Network Region 1 defines the properties for calls routed via SIP trunks to the Cisco AS5400 Universal Gateway via the Covergence CXC Session Manager. Using the change ip-network-region 1 command, enter: Name: a descriptive string such as Covergence SIP Trks Authoritative Domain: the SIP domain for this network region, e.g.,customer.com. Codec Set: the value 1 corresponding to the ip-codec-set defined in Section 3.1.3. Intra-region IP-IP Direct Audio: the value yes (the default). Inter-region IP-IP Direct Audio: the value yes (the default).
change ip-network-region 1 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: customer.com Name: Covergence SIP Trks 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: 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 Page 1 of 19
Figure 8: IP Network Region 1 Page 1

3.1.6. Define SIP Trunk Groups

Two SIP trunk groups are defined for calls to the Cisco AS5400 Universal Gateway. Each SIP trunk group requires a corresponding SIP signaling group to define the characteristics of the signaling relationship with the Covergence CXC Session Manager. Each signaling group uses a separate C-LAN card for redundancy and capacity purposes. All incoming calls use the round robin load balancing feature of the Covergence CXC Session Manager to uniformly distribute calls across both C-LANs (and thus both SIP trunk groups). If the Covergence CXC Session Manager detects a failure of SIP signaling to one C-LAN, it automatically routes all calls to the remaining C-LAN interface until the failure is corrected. All outbound calls are routed to the SIP trunk groups using Automatic Route Selection. The route patterns selected by ARS overflow from the first choice SIP trunk group to the second when a signaling failure or all-trunk-busy condition occurs. 3.1.6.1 Establish the SIP Signaling Groups Using the add signaling-group n command (where n is the number of the signaling group), configure signaling groups 11 and 12 as follows: Group Type: set to sip. Transport Method: set to tcp1. Near-end Node Name: set to the C-LAN node name (defined in Section 3.1.2) used for the respective signaling group. In these Application Notes, clan_01a02 and clan_01a03 are used for signaling group 11 and 12, respectively. Far-end Node Name: set to the interface on the Covergence CXC Session Manager that will receive the SIP signaling messages. In these Application Notes, Covergence will be used for both signaling groups 11 and 12, and corresponds to the virtual IP address of the Covergence CXC Session Manager High Availability cluster. This will be the destination IP address where SIP messages are sent. Near-end Listen Port: set to 5060, the default port of SIP signaling using tcp transport. Far-end Listen Port: set to 5060. Far-end Network Region: set to 1, the network region defined for calls using the Cisco AS5400 Universal Gateway. Far-end Domain: set to IP address or domain name of the Covergence CXC Session Manager interface used by Avaya Communication Manager. In these Application Notes, customer.com is used. Direct IP-IP Audio Connections: set to y, indicating the RTP paths should be optimized to reduce the use of media processing resources when possible.
Note that init SAT login privileges are required to change this field. Although not tested in this configuration, the Covergence CXC Session Manager supports TLS (the default), so that this restriction may not apply if the proper certificates are installed.
DTMF over IP: set to rtp-payload. This value enables Avaya Communication Manager to send DTMF transmissions using RFC 2833 [9].
The default values for the other fields may be used. The resulting form for signaling group 11 is shown below.
add signaling-group 11 SIGNALING GROUP Group Number: 11 Group Type: sip Transport Method: tcp Page 1 of 1
Near-end Node Name: clan_01a02 Near-end Listen Port: 5060 Far-end Domain: customer.com
Far-end Node Name: Covergence Far-end Listen Port: 5060 Far-end Network Region: 1
Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Enable Layer 3 Test? n Session Establishment Timer(min): 3 Direct IP-IP Audio Connections? y IP Audio Hairpinning? n
Figure 9: Signaling Group 11 The resulting form for signaling group 12 is shown below.
add signaling-group 12 SIGNALING GROUP Group Number: 12 Group Type: sip Transport Method: tcp Page 1 of 1
Near-end Node Name: clan_01a03 Near-end Listen Port: 5060 Far-end Domain: customer.com
Far-end Node Name: Covergence Far-end Listen Port: 5060 Far-end Network Region: 1
Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Enable Layer 3 Test? n Session Establishment Timer(min): 3 Direct IP-IP Audio Connections? y IP Audio Hairpinning? n
Figure 10: Signaling Group 12
3.1.6.2 Establish SIP Trunk Groups Using the add trunk-group n command (where n is the number of the trunk group), configure trunk groups 11 and 12. On Page 1 of the Trunk Group form: Group Type: set to sip. Group Name: enter a descriptive string such as SIP Trk1 to Covergence and SIP Trk2 to Covergence for trunk groups 11 and 12, respectively. TAC: enter a trunk access code such as 111 and 112 for trunk groups 11 and 12, respectively. Service Type: set to tie. Signaling Group: set to 11 and 12 (for trunk groups 11 and 12, respectively) as defined within Section 3.1.6.1. Number of Members: set to the maximum number of simultaneous calls permitted for each trunk group. Within these Application Notes, 25 was used for each trunk group. The default values may be used on the remaining pages of the trunk group form. The resulting form for trunk group11 is shown below.
add trunk-group 11 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 11 Group Type: SIP Trk1 to Covergence COR: two-way Outgoing Display? n 0 tie Auth Code? sip CDR Reports: y 1 TN: 1 TAC: 111 n Night Service: n Signaling Group: 11 Number of Members: 25 Page 1 of 21
Figure 11: Trunk Group 11 Page 1 On Page 2 of the form, set Preferred Minimum Session Refresh Interval to its maximum value of 1800 seconds. Note: This is strongly recommended to optimize Avaya Communication Manager performance during Covergence server failover conditions.
add trunk-group 11 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 1800 Page 2 of 21
Figure 12: Trunk Group 11 Page 2
The resulting forms for trunk-group 12 are shown below.

add trunk-group 12 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 12 Group Type: SIP Trk2 to Covergence COR: two-way Outgoing Display? n 0 tie Auth Code? sip CDR Reports: y 1 TN: 1 TAC: 112 n Night Service: n Signaling Group: 12 Number of Members: 25 Page 1 of 21

add trunk-group 12 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 1800 Page 2 of 21
3.1.7. Configure Calling Party Number Information

The calling party number (e.g., 18002226301) is sent in the userinfo portion of the SIP From header as shown below.
From: "Jane Smith" <sip:18002226301@customer.com>;tag=80f839da25c3db
The public-unknown-numbering command controls the calling party number sent in the SIP From field for calls originating from Avaya Communication Manager. The public-unknownnumbering feature is configured to send an 11 digit number consisting of 1800222 plus the 4 digit extension number. In these Application Notes, extensions use numbers between 6000 and 6999. Using the change public-unknown-numbering n command (where n is the leading digit of the extension range), specify the calling party number information as follows: Ext Len: set to 4, the length of the extensions used. Ext Code: set to the leading digit of the extension used. In the sample configuration, 6 is entered to cover all possible extensions between 6000 and 6999. Trk Grp(s): by default, leave blank to perform the same conversion across all trunk groups.
CPN Prefix: set to the leading digits (e.g., 1800222) that are to be sent as the calling party number. Total CPN Len: set to the total length (e.g., 11) of the calling party number to be sent. The extension number will be appended to the CPN Prefix to form the complete calling party number.
The completed public-unknown-numbering form is shown below.

change public-unknown-numbering 4 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: 1 4 6 1800222 11 Maximum Entries: 9999
Figure 15: Public Unknown Numbering
3.1.8. Configure Call Routing

3.1.8.1 Outbound Calls ARS is used to route outbound calls via the SIP trunk groups to the Covergence CXC Session Manager (that in turn routes the calls to the Cisco AS5400 Universal Gateway). The ARS route patterns support alternate routing (via the second SIP trunk group) if the primary trunk group becomes unavailable. Configuration of one outbound calling pattern supporting calls to 1-733xxx-xxx is shown in this section. Routing will select SIP trunk group 11 as the first choice, with overflow to SIP trunk group 12 as required. Further information on ARS administration is discussed in References [1] and [3]. After verifying the availability of ARS as shown in Section 3.1.1, use the change dialplan analysis command to create a feature access code (FAC) for ARS use. Dialed String: enter 9 that will become the user dialed prefix for outbound calls. Total Length: enter 1 as the length of the prefix. Call Type: enter fac as the type of prefix.
change dialplan analysis DIAL PLAN ANALYSIS TABLE Percent Full: Dialed String 0 4 5 6 9 * # Total Length 1 4 4 4 1 3 4 Call Type attd ext ext ext fac dac dac Dialed String Total Call Length Type Dialed String Total Call Length Type 1 Page 1 of 12
Figure 16: Dial Plan Analysis
Use the change feature-access-codes command to assign the feature access code 9 to Auto Route Selection (ARS) - Access Code 1 as shown below.
change feature-access-codes FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: Abbreviated Dialing List2 Access Code: Abbreviated Dialing List3 Access Code: Abbreviated Dial - Prgm Group List Access Code: Announcement Access Code: *71 Answer Back Access Code: Auto Alternate Routing (AAR) Access Code: Auto Route Selection (ARS) - Access Code 1: 9 Automatic Callback Activation: Call Forwarding Activation Busy/DA: *61 All: *62 Call Forwarding Enhanced Status: Act: Call Park Access Code: Call Pickup Access Code: CAS Remote Hold/Answer Hold-Unhold Access Code: CDR Account Code Access Code: Change COR Access Code: Change Coverage Access Code: Contact Closure Open Code: Page 1 of 7
Access Code 2: Deactivation: Deactivation: *60 Deactivation:
Close Code:
Figure 17: ARS Feature Access Code Use the change ars analysis nn command to configure the ARS route pattern selection rules as follows. Here nn is 17, the first two digits of the dialed number after the ARS access code. Dialed String: enter the leading digits (e.g., 1733) necessary to uniquely select the desired route pattern. Total Min: enter the minimum number of digits (e.g., 11) expected for this PSTN number. Total Max: enter the maximum number of digits (e.g., 11) expected for this PSTN number. Route Pattern: enter the route pattern number (e.g., 11) to be used. The route pattern (to be defined next) will specify the trunk group(s) to be used for calls matching the dialed number. Call Type: enter fnpa, the call type for North American 1+10 digit calls.
change ars analysis 17 ARS DIGIT ANALYSIS TABLE Location: all Dialed String 1733 Total Min Max 11 11 Route Pattern 11 Call Type fnpa Node Num Page 1 of 1 2
Percent Full: ANI Reqd n
Figure 18: ARS Digit Analysis Entries
Use the change route-pattern n command (where n is the Route Pattern number used above) to specify the SIP trunk groups selected for the outbound call. In the form: Pattern Name: enter a descriptive string such as Covergence11/12 to describe the routing pattern. Secure SIP?: leave as n, the default. Grp No: enter the trunk groups to be used in priority order. In this configuration, trunk group 11 is the first choice route followed by trunk group 12. FRL: enter the minimum facility restriction level (e.g., 1) necessary to use this trunk group, with 0 being the least restrictive. The FRL within the Class of Restriction (COR) assigned to the station must be greater than or equal to 1 in this case to use these trunk groups. Pfx Mrk: enter 1, to always send the prefix 1 on 10 digit calls. LAR: enter the routing behavior to be followed if the call is not successfully completed using the trunk group. Next will cause the call to attempt to use the next choice in the routing pattern. None indicates that no further attempts will be made to complete the call. In the example below, a call that fails when attempting to use trunk group 11, will automatically attempt to use trunk group 12 before being abandoned. The default values for the remaining fields may be used. The completed route pattern form is shown below.
change route-pattern 11 Pattern Number: 11 Grp FRL NPA Pfx Hop Toll No. No Mrk Lmt List Del Dgts 1: 11 1 1 2: 12 1 1 3: 4: 5: 6: BCC VALUE TSC CA-TSC 0 1 2 M 4 W Request 1: 2: 3: 4: 5: 6: y y y y y y y y y y y y y y y y y y y y y y y y y y y y y y n n n n n n n n n n n n Page Pattern Name: Covergence11/12 Secure SIP? n Inserted Digits 1 of 3
DCS/ QSIG Intw n n n n n n
IXC
user user user user user user
ITC BCIE Service/Feature PARM
rest rest rest rest rest rest
No. Numbering LAR Dgts Format Subaddress next none none none none none
Figure 19: Route Pattern 11
3.1.8.2 Incoming Calls This step configures the routing of incoming DID numbers to the proper extensions. In these Application Notes, the following incoming toll-free 800 numbers are used. Digits Received (within SIP INVITE message) 800 222 7001 800 222 7002 800 222 7003 800 222 7006 Extension (or Hunt Group) Answering 6301 (Avaya 4621SW IP Telephone) 6302 (Avaya 4621SW IP Telephone) 6303 (Avaya 6424D Telephone) Forwarded to PSTN 1 733 333 2226 via SIP trunk
Table 2 Incoming Call Routing Use the change inc-call-handling-trmt trunk-group n command (where n is the SIP trunk group number) to administer the incoming number routing. This administration must be done for each incoming trunk group. Called Len: enter the total number of incoming digits received (e.g., 10). Called Number: enter the specific digit pattern to be matched. Del: enter the number of leading digits that should be deleted Insert: enter the specific digits to be inserted at the beginning of the adjusted incoming digit string, forming the complete number. The completed inc-call-handling-trmt form for trunk group 11 is shown below. The form for trunk group 12 is identical.
change inc-call-handling-trmt trunk-group 11 INCOMING CALL HANDLING TREATMENT Service/ Called Called Del Insert Feature Len Number tie 10 8002227006 10 917333332226 tie 10 80022270 8 63 Page 1 of 30
Figure 20: Incoming Call Treatment for Trunk 11
3.1.9. Save Avaya Communication Manager Changes

This completes the configuration of Avaya Communication Manager. Use the save translation command to make the changes permanent.
4. Configure the Covergence CXC Session Manager

This section describes the configuration of the Covergence CXC Session Manager, which acts as an intermediary between the Avaya Communication Manager C-LAN interfaces and the Cisco AS5400 Universal Gateway. The following subsections show the web interface screens that highlight the important aspects relative to the sample SIP trunking configuration. The relevant link on the left side is circled, corresponding to the configuration screen shown on the right. The entire CLI-based configuration file is included in Appendix A for reference.
These Application Notes assume the Covergence CXC Session Manager has been previously installed in an HA configuration according Covergence guidelines, and with the IP addresses indicated for it in Figure 1. Basic installation and HA configuration is standard and beyond the scope of this SIP trunking application.
4.1. Physical and Network Interfaces

In the sample configuration, the 10.1.1.0/24 subnet was configured as the private side of the Covergence CXC Session Manager. This subnet had IP routing connectivity with the 192.168.99.0/24 subnet used by Avaya Communication Manager IP interfaces, as shown in Figure 1. All SIP signaling and RTP media sent by Avaya Communication Manager were routed via the Covergence CXC Session Manager to the Cisco AS5400 Universal Gateway. The Covergence CXC Session Manager was configured with a Virtual Router Redundancy Protocol (VRRP) virtual IP address 10.1.1.100, corresponding to the active server in the HA cluster. This IP address was used in all Avaya Communication Manager administration. The web browser screen for this virtual interface is shown below. Note that the C-LAN IP addresses for the Avaya Communication Manager SIP trunks are specified in the ip section under trusted-peer. This allows signaling from these interfaces without authentication challenges.
Figure 21: Private Side Virtual IP Address

Ethernet interface eth1, supported on each Covergence CXC Session Manager server in the redundant configuration, is dedicated for the 10.1.1.0/24 LAN subnet, and has its own physical IP address. 10.1.1.101 is configured for box CXC1 and 10.1.1.102 for box CXC2. The configuration for box CXC1 is shown below. The configuration for box CXC2 is identical except for the IP address value.
Figure 22: Private Side (eth1) Physical Interface Configuration
4.2. Default Session Configuration

The default session configuration defines the signaling characteristics of the SIP interfaces on the Covergence CXC Session Manager. In the sample configuration, the default settings were used, with the following exceptions: Under sip-directive, directive should be set to allow (Figure 23). Addition of the multicast address field (maddr) in the SIP Contact header must be disabled in order to guarantee successful SIP calls to/from Avaya Communication Manager (Figure 24). Under contact-uri-settings-in-leg, add-maddr must be set to disabled. Accounting should be done locally. Under accounting, target should be set to database vsp/accounting/database/group local. (Figure 25). The capability to anchor the media should be enabled; i.e., RTP passes through the Covergence CXC Session Manager (Figure 26). Under media, anchor should be set to enabled.
Figure 23: sip-directive Configuration
Figure 24: Contact Header Configuration
Figure 25: Call Accounting Configuration
Figure 26: Media Anchoring Configuration
4.3. SIP Trunks

The SIP domain of the customer configuration is defined under the static-stack-settings section. Enter customer.com in the domain-name field.
Figure 27: SIP Domain Name
The Avaya Communication Manager SIP signaling groups and the Cisco AS5400 Universal Gateway will send SIP messages to the SIP server interfaces defined below under the enterprise/servers section of the administration GUI. Add a sip-gateway for each C-LAN interface and Cisco AS5400 Universal Gateway used for SIP trunks. In the sample configuration, two servers were added for C-LANs 1a02 and 1a03, and one server was added for the Cisco AS5400 Universal Gateway. Figures 28-29 show the relevant parts of the configuration screen for adding the server for C-LAN 1a02. Fill in the name and domain fields when adding the server. The GUI wizard will step through the sip-gateway screens. Fill in the name and domain fields in the sip-gateway screen (Figure 28) and then click on Finish at the bottom of the screen (not shown).
Figure 28: Adding a sip-gateway (Part 1)
Under other properties on the new sip-gateway screen, click on the Configure link next to server-pool (Figure 29).
Figure 29: Adding a sip-gateway (Part 2)
Click on Add server and Add handle-response to define the trunk interface characteristics. The resulting configuration is shown in Figure 30. When adding the server, specify the server name (CLAN 02), the IP address of the C-LAN in host (192.168.99.52), and set transport to TCP. Use Add handle-response to define the appropriate SIP response codes which will cause the INVITE to be re-tried using the next server interface in the hunt group (defined in the next section). Default values can be used for the remaining fields.
Figure 30: Adding the server-pool
Repeat the above steps for C-LAN 1a03 and the Cisco AS5400 Universal Gateway. In the sample configuration, the transport parameter was set to UDP for the server corresponding to the Cisco AS5400 Universal Gateway. The result of adding all servers is shown in Figure 31.
Figure 31: Final Enterprise Servers (SIP Trunks) Screen
4.4. Load Balancing

Load balancing is accomplished by defining a hunt-group under the carriers section of the GUI. Click on carriers on the left side, and then Add hunt-group on the right side. Set the name (Hunt CLAN 02 and CLAN 03), the call-hunting-type to sequential, and click on Add option (not shown) to add each server defined in the previous section that is to be a member of this hunt group (CLAN 02 and CLAN 03 in the sample configuration). The result for the sample configuration is shown below.
Figure 32: Hunt Group Definition
4.5. SIP Routing

Call routing configuration is accomplished in the dial-plan section of the GUI. In the sample configuration, source based routing was used to route all calls coming from Avaya Communication Manager to the Cisco AS5400 Universal Gateway, and vice versa. Click on dial-plan on the left side and then scroll to the source-route section on the right side. Click on Add source route (not shown). Figures 33-34 show the configuration for the source route from C-LAN 1a02 to the Cisco AS5400 Universal Gateway. Assign a name. In the source-match section, set the type field to server and select the sip-gateway corresponding to C-LAN 1a02 in the source-server field. Similarly in the peer section, select the type and server corresponding to the Cisco AS5400 Universal Gateway. In the alter-request-uri and alter-to-uri sections, set the type field to next-hop-ip, and in the alter-from-uri section, set type to local-ip.
Figure 33: Source Route from C-LAN 1a02 to Cisco AS5400 Universal Gateway (Part 1)
Figure 34: Source Route from C-LAN 1a02 to Cisco AS5400 Universal Gateway (Part 2)
Repeat the above steps for the source route from C-LAN 1a03 to the Cisco AS5400 Universal Gateway. For the source route from the AS5400 Universal Gateway to Avaya Communication Manager, calls should be routed to the hunt group containing both C-LANs. Set the type field in the peer section to hunt-group and the hunt-group field to the hunt group defined in Section 4.4, as shown in Figure 35.
Figure 35: Source Route from Cisco AS5400 Universal Gateway to C-LAN Hunt Group (Part 1)
In the alter-request-uri section, set type to host and host-name to customer.com".
Figure 36: Source Route from Cisco AS5400 Universal Gateway to C-LAN Hunt Group (Part 2)
Figure 37 shows all of the routes for the sample configuration.
Figure 37: All Source Routes
4.6. Saving Configuration Changes

Configuration changes can be saved by clicking on the Set button at the top of any of the configuration screens (see Figure 38). To activate these changes in the running system, click on Configuration in the upper left corner and select Update and save configuration from the drop-down menu. Click on OK and Yes in the following two dialog boxes, and the confirmation screen will be displayed (Figure 39).
Figure 38: Saving Configuration Changes (Part 1)
Figure 39: Saving Configuration Changes (Part 2)
5. Verification Steps
This section provides steps that may be performed to verify the operation of the direct SIP trunking configuration described in these Application Notes. Avaya Communication Manager list trace station, list trace tac, status station and/or status trunk-group commands are helpful diagnostic tools to verify correct operation and to troubleshoot problems. Also using a SIP protocol analyzer such as WireShark (a.k.a., Ethereal) to monitor the SIP messaging at the various interfaces (C-LAN, Covergence CXC Session Manager and/or Cisco AS5400 Universal Gateway) is often very helpful in troubleshooting issues.
The Covergence CXC Session Manager also has call tracing and event logs that are useful in troubleshooting. Under default-session-config/log-alert, enable the alert, logging, messagelogging, tracing, and message-auditing parameters as indicated in Figure 40. These parameters should be enabled for diagnostic purposes only, since system performance is affected.
Figure 40: Logging/Tracing Configuration
Call flows can be traced by clicking on the Call Logs tab in the web interface GUI (Figure 41). Click on Refresh to update the log after a call scenario has been attempted. Click on the Call Diagram link corresponding to the top call flow to see a graphical depiction of the signaling (shown in Figure 42).
Figure 41: Call Tracing in Covergence CXC Session Manager Session Display
Figure 42: Call Tracing in Covergence CXC Session Manager Call Sequence Diagram A more detailed message trace can also be viewed and/or saved by clicking on the Download messages as text link. The resulting display window is shown in Figure 43.
Figure 43: Call Tracing in Covergence CXC Session Manager Call Trace Text Display The above diagnostic tools can be used to troubleshoot issues encountered while executing the following functional scenarios: Incoming Calls Verify that calls placed from a PSTN telephone to the DID number assigned are properly routed via the SIP trunk group(s) to the expected telephone, hunt group, etc. Verify the talk-path exists in both directions, that calls remain stable for several minutes and disconnect properly.
Outbound Calls Verify that calls placed to a PSTN telephone are properly routed via the SIP trunk group(s) defined in the ARS route patterns. Verify that the talk-path exists in both directions and that calls remain stable and disconnect properly. Direct IP-IP Connections This applies if IP telephones and Direct IP-IP are used. Verify that stable calls are using Direct IP-IP talk paths using the status station or status trunk-group commands. When Direct IP-IP is used, the Audio Connection field will indicate ip-direct instead of ip-tdm. Load Balancing of Incoming Calls This applies if multiple SIP trunk groups (using multiple C-LANs and Covergence CXC-550 Session Manager Load Balancing) are used. Verify that incoming calls are distributed across the trunk groups defined. Alternate Routing of Inbound Calls on C-LAN failure This applies if multiple SIP trunk groups (using multiple C-LANs) are used. Maintenance busy the C-LAN associated with an incoming SIP trunk group and verify using the list trace station or list trace trunk commands that inbound calls are routed to the active SIP trunk group (using a separate C-LAN). Verify that the original trunk group is used once the C-LAN is returned to service. Repeat for other incoming SIP trunk groups. Note: This may be service affecting! Alternate Routing of Outbound Calls on C-LAN failure This applies if multiple SIP trunk groups (using multiple C-LANs) are used. Maintenance busy the C-LAN associated with the first-choice trunk group and verify using the list trace station or list trace trunk commands that outbound calls overflow to the next SIP trunk group (using a separate C-LAN). Verify that the original trunk group is used once the C-LAN is returned to service. Repeat for other route-patterns using these trunk groups. Note: This may be service affecting!
6. Support
For technical support on the Covergence CXC Session Manager, visit support.covergence.com.
7. Conclusion
These Application Notes describe the configuration steps required to establish SIP trunking directly with Avaya Communication Manager to Covergence CXC Session Manager and a Cisco AS5400 Universal Gateway for the purpose of PSTN interconnection. This configuration was successfully tested with the demonstration of calls in both directions with the PSTN. The ability to use incoming load balancing across multiple Avaya Communication Manager C-LAN interfaces and endure a C-LAN interface isolation or failure was shown.
8. References
The Avaya product documentation is available at http://support.avaya.com. [1] Administrator Guide for Avaya Communication Manager, February 2007, Issue 3, Document Number 03-300509. [2] Adding New Hardware for Avaya Media Servers and Gateways, February 2007, Issue 2, Release 4.0, Document Number 03-300684. [3] Feature Description and Implementation for Avaya Communication Manager, Issue 5, Document Number 555-245-205. [4] SIP Support in Avaya Communication Manager Running on the Avaya S8300, S8400, S8500 series and S8700 series Media Server, March 2007, Issue 6.1, Document Number 555-245-206. [5] 4600 Series IP Telephone Release 2.6 LAN Administrator Guide, August 2006, Issue 4, Document Number 555-233-507. The following documentation is provided with the Covergence CXC Session Manager or is available from Covergence Technical Support. See support.covergence.com for further information. [6] Eclipse SIP Security and Management Solutions System Administration Guide, Release 3.2, 780-0003-00, Revision 03.02.2007, June, 2007. [7] Eclipse SIP Security and Management Solutions Session Services Configuration Guide, Release 3.2, 780-0001-00, Revision 03.02.2007, June, 2007. Several Internet Engineering Task Force (IETF) standards track RFC documents were referenced within these Application Notes. The RFC documents may be obtained at: http://www.rfc-editor.org/rfcsearch.html. [8] RFC 3261 - SIP (Session Initiation Protocol), June 2002, Proposed Standard [9] RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, May 2000, Proposed Standard
Appendix A Command Line Interface Configuration of Covergence CXC Session Manager

# # Copyright (c) 2004-2007 Covergence Inc. # All Rights Reserved. # # File: /cxc/cxc.cfg # Date: 16:14:34 Thu 2007-10-11 # config cluster set name UHG config box 1 set hostname CXC1 set name CXC1 set contact "" set location "" set identifier 00:04:23:b5:3f:e2 config interface eth0 set speed 100Mb set autoneg disabled config ip Heartbeat_cxc1 set ip-address static 192.168.1.1/24 set classification-tag "" config ssh return config bootp-server return config icmp return config vrrp return config messaging return return return config interface eth1 set speed 10Mb set autoneg disabled config ip static set ip-address static 10.1.1.101/24 set classification-tag "" config ssh return config web set protocol https 443 0 return config icmp return return return config interface eth2 set speed 10Mb set autoneg disabled config ip static FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 41 of 51 CXC550_SIP_PSTN
set ip-address static 142.16.58.101/24 set classification-tag "" config ssh return config web set protocol https 443 0 return config icmp return return return config cli set prompt CXC_Box-1 set banner "" return config console return return config box 2 set hostname CXC2 set name CXC2 set contact "" set location "" set identifier 00:0e:0c:e9:f3:f0 config interface eth0 set speed 100Mb set autoneg disabled config ip Heartbeat_cxc2 set ip-address static 192.168.1.2/24 set classification-tag "" config ssh return config icmp return config vrrp return config messaging return return return config interface eth1 set speed 10Mb set autoneg disabled config ip static set ip-address static 10.1.1.102/24 set classification-tag "" config ssh return config web set protocol https 443 0 return config icmp return return return FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 42 of 51 CXC550_SIP_PSTN
config interface eth2 set speed 10Mb set autoneg disabled config ip static set ip-address static 142.16.58.102/24 set classification-tag "" config ssh return config web set protocol https 443 0 return config icmp return return return config cli set prompt CXC_Box-2 set banner "" return config console return return set share-signaling-entries true set mirror-media-streams true config vrrp config vinterface vx2 set group 1 set host-interface cluster\box 1\interface eth2 set host-interface cluster\box 2\interface eth2 config ip "Public Side" set ip-address static 142.16.58.100/24 set classification-tag "" config ssh return config web set protocol https 443 0 return config sip set udp-port 5060 "" "" any 0 set tcp-port 5060 "" "" any 0 set tls-port 5061 "" "" any 0 return config icmp return config media-ports return config routing config route Net_142 set destination network 142.16.58.0/24 set gateway 142.16.58.1 return config route Cisco set destination host 142.16.57.2 set gateway 142.16.58.1 return FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 43 of 51 CXC550_SIP_PSTN
config route bogus set destination host 5.5.5.5 set gateway 142.16.58.1 return return return return config vinterface vx111 set group 1 set host-interface cluster\box 1\interface eth1 set host-interface cluster\box 2\interface eth1 config ip Private set ip-address static 10.1.1.100/24 set classification-tag "" set trusted-peer 192.168.99.52 set trusted-peer 192.168.99.53 config ssh return config web set protocol https 443 0 return config sip set nat-translation enabled set udp-port 5060 "" "" any 0 set tcp-port 5060 "" "" any 0 set tls-port 5061 "" "" any 0 return config icmp return config media-ports return config routing config route "CLAN 02" set destination host 192.168.99.52 set gateway 10.1.1.1 return config route "CLAN 03" set destination host 192.168.99.53 set gateway 10.1.1.1 return config route 10_net set destination network 10.1.1.0/24 set gateway 10.1.1.1 set metric 2 return return return return return return config services config event-log config file kernel set filter krnlsys debug FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 44 of 51 CXC550_SIP_PSTN
return config file access-log set filter access debug return config file eventlog set filter all error return config file testing set filter sip info set size 3 return config file "System Info" set filter system info return config local-database set filter dosDatabase info return config smtp mail.uhg set admin disabled set destination-mailbox bill.uhg.net set reply-mailbox covergence@uhg.net set filter access info set filter all error return return return config master-services config cluster-master set host-box cluster\box set host-box cluster\box set group 1 return config accounting set host-box cluster\box set host-box cluster\box set group 1 config settings return return config database set host-box cluster\box set host-box cluster\box set group 1 set preempt true set media enabled return config call-failover set host-box cluster\box set host-box cluster\box set group 1 set takeover-timer-value return return
1 2
1 2
1 2
1 2 1000
config vsp set admin enabled set local-identity "" set local-normalization disabled set registration-proxy enabled config registration-service return config default-session-config config sip-settings set route-hdr request set handle-3xx-locally enabled set preserve-3xx-contact enabled set set-3xx-contact-host "" set to-header-follows-contact-header enabled return config sip-directive set directive allow return config contact-uri-settings-in-leg set add-maddr disabled return config accounting set target database "vsp\accounting\database\group local" set header Call-ID set report-failed-calls enabled return config media set anchor enabled config nat-traversal set symmetricRTP true return config recording-policy set record enabled return set auto-conference disabled "" out set introduction "" set music-on-hold "" set inactivity-timeout enabled "0 days 01:00:00" set rtp-stats enabled return config log-alert set alert enabled "services\event-log\local-database" info set logging enabled set tracing enabled set message-auditing enabled error return config forking-settings set forking-type sequential return return config static-stack-settings set domain-name customer.com set domain-alias test.uhg.com set domain-alias 142.16.58.100 set location-lookup-pattern user-only FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 46 of 51 CXC550_SIP_PSTN
set t1 200 set t2 1100 return set call-admission-control disabled set registration-admission-control disabled set call-response-string-at-threshold "" config session-config-pool config entry "codec test" config sip-directive set directive allow return config in-codec-preferences return config out-codec-preferences set preference audio pcma 0 set preference audio iLBC 1 set preference audio pcmu 2 set preference audio speex 3 return return return config dial-plan config arbiter Weighted set call-hunting-type sequential set call-routing-on request-uri set rule weighted-call-average set subscriber-match phone-prefix 800 1 set priority 50 return config route test set admin disabled set request-uri-match phone-prefix 800 1 set location-match-preferred no set peer hunt-group "vsp\carriers\hunt-group ""Hunt CLAN 02 set alter-request-uri host customer.com set alter-to-uri next-hop-ip set alter-from-uri local-ip return config source-route "From CLAN 02 to Cisco PSTN" set source-match server "vsp\enterprise\servers\sip-gateway set location-match-preferred no set peer server "vsp\enterprise\servers\sip-gateway ""Cisco set alter-request-uri next-hop-ip set alter-to-uri next-hop-ip set alter-from-uri local-ip return config source-route "From CLAN 03 to Cisco PSTN" set source-match server "vsp\enterprise\servers\sip-gateway set location-match-preferred no set peer server "vsp\enterprise\servers\sip-gateway ""Cisco set alter-request-uri next-hop-ip set alter-to-uri next-hop-ip set alter-from-uri local-ip return config source-route "From Cisco to Hunt CLAN 02 and CLAN 03" FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved.
and CLAN 03"""
""CLAN 02""" PSTN"""
""CLAN 03""" PSTN"""
set source-match server "vsp\enterprise\servers\sip-gateway ""Cisco PSTN""" set location-match-preferred no set peer hunt-group "vsp\carriers\hunt-group ""Hunt CLAN 02 and CLAN 03""" set alter-request-uri host customer.com set alter-to-uri next-hop-ip set alter-from-uri local-ip return return config registration-plan config settings set alpha-numeric-phone-expression AA set alpha-numeric-phone-expression AB set alpha-numeric-phone-expression BB set alpha-numeric-phone-expression ^$ return config route register-732 set to-uri-match phone-prefix 732 4 set action accept set response-string "" return return config enterprise config servers config sip-gateway "CLAN 02" set peer-identity "" set domain customer.com set routing-tag "" set user "" set password-tag "" config server-pool config server "CLAN 02" set host 192.168.99.52 set transport TCP set max-number-of-concurrent-calls 5 return set handle-response 403 try-next-peer set handle-response 404 try-next-peer set handle-response 503 try-next-peer set handle-response 504 try-next-peer return return config sip-gateway "CLAN 03" set peer-identity "" set domain customer.com set routing-tag "" set user "" set password-tag "" config server-pool config server "CLAN 03" set host 192.168.99.53 set transport TCP set max-number-of-concurrent-calls 5 return set handle-response 403 try-next-peer FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 48 of 51 CXC550_SIP_PSTN
set handle-response 404 try-next-peer set handle-response 503 try-next-peer set handle-response 504 try-next-peer return return config sip-gateway "Cisco PSTN" set peer-identity "" set domain "" set routing-tag "" set user "" set password-tag "" config server-pool config server "Cisco PSTN Gwy" set host 142.16.57.2 return set handle-response 404 try-next-peer return return return return config carriers config hunt-group "Hunt CLAN 02 and CLAN 03" set option server "vsp\enterprise\servers\sip-gateway ""CLAN 02""" 1 set option server "vsp\enterprise\servers\sip-gateway ""CLAN 03""" 1 return return config accounting config database config group local config server localdb set type local set username postgres set password-tag postgres return config call-field-filter set filter session-id+recorded+call-id+to+from+method+incoming-requesturi+outgoing-request-uri+previous-hop-ip+previous-hop-via+next-hop-ip+nexthop-dn+header+origin+setup-time+connect-time+disconnect-time+disconnectcause+scp-name+call-id-2+call-type+disconnect-error-type+ani+new-ani+calldest-cr-name+cdr-type+hunting-attempts+call-pdd+called-party-after-src-dialplan+last-status-message set source-media-filter packets-received+packets-lost+packetsdiscarded+pdv+codec+latency+rfactor+mime-type+mos+ip-in+ip-out+anchor-ipin+anchor-ip-out set destination-media-filter packets-received+packets-lost+packetsdiscarded+pdv+codec+latency+rfactor+mime-type+mos+ip-in+ip-out+anchor-ipin+anchor-ip-out set source-caller-filter regid+realm-name set destination-caller-filter regid+realm-name return return return return config dns config resolver FS; Reviewed: SPOC 12/6/2007 Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved. 49 of 51 CXC550_SIP_PSTN
set admin disabled set server 2.2.2.2 UDP 53 100 ALL return return config location-service config settings set cache-poll-interval 3600 return return config database set accounting-history 12 set call-details-history 12 set media-history 12 set file-transfer-history 3 set im-history 3 return config settings set sip-stack-pre-auth-timeout 60 set sip-stack-pre-auth-max-pending 131072 set max-number-of-registrations 100000 set pending-registrations-high-watermark 500000 set pending-registrations-low-watermark 100000 set registrations-high-cpu-threshold 90 set max-retransmissions 2 set out-of-context-message-action refuse 400 set out-of-context-message-media-cleanup disabled set preserve-3xx-contact enabled set register-retransmit-detection disabled return return config external-services return config preferences config cms-preferences return return config access return config features set media-sessions 1000 set media-encryption-sessions 625 set lcs-sametime-gateway-sessions 0 return
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 Solution & Interoperability Test Lab at interoplabnotes@list.avaya.com

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Avaya Solution & Interoperability Test Lab

FS; Reviewed: SPOC 12/6/2007

FS; Reviewed: SPOC 12/6/2007

Figure 1 Direct SIP Trunking Test Configuration

2. Equipment and Software Validated

Table 1 Equipment and Versions

3. Configure Avaya Communication Manager

FS; Reviewed: SPOC 12/6/2007

3.1. SIP Trunk Configuration

Figure 2: System-Parameters Customer-Options Form Page 2

FS; Reviewed: SPOC 12/6/2007

Digital Loss Plan Modification? n DS1 MSP? n DS1 Echo Cancellation? n

Figure 3: System-Parameters Customer-Options Form Page 3

3.1.2. Determine Node Names

Figure 4: IP Node Names

FS; Reviewed: SPOC 12/6/2007

3.1.3. Define IP Codec Sets

Figure 5: IP Codec Set 1

3.1.4. Configure the C-LAN IP Network Region Assignment

C-LAN 01A02 TN799 D clan_01a02 192.168.99.52 255.255.255.0 . . . y 1 n

Link: 12 Allow H.323 Endpoints? y Allow H.248 Gateways? y Gatekeeper Priority: 5

Figure 6: IP Interface of C-LAN 1a02 used for SIP Signaling Group 11

Link: 13 Allow H.323 Endpoints? y Allow H.248 Gateways? y Gatekeeper Priority: 5

Figure 7: IP Interface of C-LAN 1a03 used for SIP Signaling Group 12

3.1.5. Define IP Network Region

Figure 8: IP Network Region 1 Page 1

3.1.6. Define SIP Trunk Groups

FS; Reviewed: SPOC 12/6/2007

Figure 10: Signaling Group 12

FS; Reviewed: SPOC 12/6/2007

Figure 12: Trunk Group 11 Page 2

FS; Reviewed: SPOC 12/6/2007

The resulting forms for trunk-group 12 are shown below.

Figure 13: Trunk Group 12 Page 1

Figure 14: Trunk Group 12 Page 2

3.1.7. Configure Calling Party Number Information

FS; Reviewed: SPOC 12/6/2007

The completed public-unknown-numbering form is shown below.

Figure 15: Public Unknown Numbering

3.1.8. Configure Call Routing

Figure 16: Dial Plan Analysis

FS; Reviewed: SPOC 12/6/2007

Access Code 2: Deactivation: Deactivation: *60 Deactivation:

Percent Full: ANI Reqd n

Figure 18: ARS Digit Analysis Entries

FS; Reviewed: SPOC 12/6/2007

DCS/ QSIG Intw n n n n n n

user user user user user user

ITC BCIE Service/Feature PARM

rest rest rest rest rest rest

Figure 19: Route Pattern 11

FS; Reviewed: SPOC 12/6/2007

Figure 20: Incoming Call Treatment for Trunk 11

3.1.9. Save Avaya Communication Manager Changes

4. Configure the Covergence CXC Session Manager

FS; Reviewed: SPOC 12/6/2007

4.1. Physical and Network Interfaces

Figure 21: Private Side Virtual IP Address

Figure 22: Private Side (eth1) Physical Interface Configuration

FS; Reviewed: SPOC 12/6/2007

4.2. Default Session Configuration

Figure 23: sip-directive Configuration