CCO/CBC IP Office with AT&T IP Flex SIP Trunk

[tmnt1981]

Hello all,

Has anyone successfully implemented an IP Office with an AT&T IP Flex SIP Handoff with no SBC?
I have been working on this with no success. I followed the devconnect document for 9.0 (

https://www.devconnectprogram.com/fileMedia/download/a00e94f3-0f07-4913-9ac2-6bd83419e63c)

but am getting "The number you have dialed is being checked for trouble," on incoming and "Waiting for Line" on outgoing.
The SIP hand off was originally on the same as the data hand off (int ge0/, on the Cisco Gateway) but I had AT&T move it to a separate interface to isolated the end users firewall. I am getting no support from Avaya since it is rel 9.0 and am getting some pushback from the end user regarding an upgrade. They are not down as this was a PRI to SIP transition and since we cannot get calls across on the SIP trunk they are still using the PRI. Any assistance would be greatly appreciated

 

[rdoubrava]

Are you using a firewall?

 

[tmnt1981]

Initially they were, but since AT&T moved the sip handoff to a separate interface we are bypassing the firewall.

 

[tmnt1981]

Here is a trace in SysMonitor
17:40:51 513456322mS SIP Call Tx: 17
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKf4ffc6ababc5966c61ee4c58cbbffc45
From: "Extn362" <sip:332@12.16.193.41>;tag=b403bca4028036f5
Call-ID: 1d791d0a6a685c35b71728e0dbe96d9a
CSeq: 1146764784 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer
Server: IP Office 9.0.2.0 build 860
To: <sip:18474272350@12.16.193.41>;tag=b403bca4028036f5
Content-Length: 0

17:40:51 513456322mS SIP Tx: UDP 192.168.1.2:5060 -> 192.168.1.2:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKf4ffc6ababc5966c61ee4c58cbbffc45
From: "Extn362" <sip:332@12.16.193.41>;tag=b403bca4028036f5
Call-ID: 1d791d0a6a685c35b71728e0dbe96d9a
CSeq: 1146764784 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer
Server: IP Office 9.0.2.0 build 860
To: <sip:18474272350@12.16.193.41>;tag=b403bca4028036f5
Content-Length: 0

17:40:51 513456324mS SIP Rx: UDP 192.168.1.2:5060 -> 192.168.1.2:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bKf4ffc6ababc5966c61ee4c58cbbffc45
From: "EXT362" <sip:332@12.16.193.41>;tag=b403bca4028036f5
Call-ID: 1d791d0a6a685c35b71728e0dbe96d9a
CSeq: 1146764784 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer
Server: IP Office 9.0.2.0 build 860
To: <sip:18474272350@12.16.193.41>;tag=b403bca4028036f5
Content-Length: 0

17:40:51 513456452mS PRN: Monitor Status IP 500 V2 9.0.2.0 build 860
17:40:51 513456452mS PRN: LAW=U PRI=1, BRI=0, ALOG=8, VCOMP=64, MDM=0, WAN=0, MODU=0 LANM=0 CkSRC=1 VMAIL=1(VER=3 TYP=1) 1-X=0 CALLS=1(TOT=972)
17:40:52 513457584mS H323Evt: Recv: RegistrationRequest 10.1.7.80; Endpoints registered: 52; Endpoints in registration: 0
17:40:52 513457987mS CMMap: PCG::UnmapBChan pcp[924]b0r1 cp_b 0 other_cp_b 0

********** Warning: Logging to Screen Stopped **********

 

[janni78]

Why is it sending the SIP messages to itself? (SIP Rx: UDP 192.168.1.2:5060 -> 192.168.1.2:5060)
I think the issue is with you line config.

"Trying is the first step to failure..." - Homer

 

[tmnt1981]

Thanks janni78, that is what another engineer I work with said. He has had success with IP flex with Mitel systems and he said the same thing. Thats a mistake on my part. I'm contacting the customer and will make the change. It should be 192.168.1.1 and 192.168.1.2 is the address for Lan 2. I will post the results but I thank you for the response.