Networking M1 to Adtran NetVanta 7100 thru NRS

[LuckyThirteen]

Hello,
I am in the process of attempting to network a CS1000M Rls 5.5 PBX to an Adtran NetVanta 7100, using the NRS. I have gotten the NetVanta to register to the NRS as a SIP gateway endpoint, and built my routing entries on both sides. When a call is placed from the NetVanta phone to a DN (x6098)that is a routing entry for the M1, the call fails. The NetVanta tech that I am working with can see the SIP messages being exchanged. The NetVanta and the M1 are talking to each other, but he is seeing some invalid SIP messages from the M1 side. Instead of seeing "6098@adtransip.com", he is seeing it as "6098.udp.cdp@adtransip.com". Adtransip.com is the SIP domain name.

Has anyone networked these two together, or know what I can change in the NRS to get rid of the .udp.cdp? Or, am I missing something else all together? Thanks in advance.

 

[allenmac]

I think the Adtran should be in the same SIP domain as the rest of the system. That would be the first thing I looked at.

 

[LuckyThirteen]

The Adtran is in the same SIP domain, and is even registered dynamically as an endpoint. I tried changing the DN type of the routing entry 6098 from E164 Subscriber to CDP, and also to Private Special, and nothing changed. I also went into the SIP URI map of the Node in Element Manager and took "cdp.udp" out of the Private/CDP domain name, also nothing changed.

 

[allenmac]

sorry, I misread the post.

I use AudioCodes MediaPack gateways, and here is a set of SIP messaging from a call initiated from the Audiocodes to My 1140e desk phone. These messages start with the initial response from my NRS to the call request from the Audiocodes.

As you can see the Audiocodes sip context is: "sip:6489@MySIPDomain.com", and my deskphone's sip context is: "sip:6888;phone-context=ChevyChase.Corporate@MySIPDomain.com", so it doesn't seem like the SIP message above is really very troubling at all.

In my experience, it is fairly difficult to get the configuration correct. I struggled with getting the AudioCodes configured, and found a thread on this board by Pronei that was very helpful.

My gut feeling is that there is one or more options not configured correctly in the Adtran device that is causing your issue.

26d:13h:1m:37s ( lgr_flow)(116217 ) ---- Incoming SIP Message from 172.20.1.20:5060 ---- [Time: 13:1:37]

26d:13h:1m:37s SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.110.31.251;branch=z9hG4bKac1020794782;received=10.110.31.251 From: "AudioCodes 2" <sip:6489@MySIPDomain.com>;tag=1c1020792010 To: <sip:6888@MySIPDomain.com>;tag=31942 Call-ID: 1020791712266200913137@10.110.31.251 CSeq: 1 INVITE Contact: <sip:6888;phone-context=ChevyChase.Corporate@MySIPDomain.com:5060;maddr=172.20.103.253;transport=udp;x-nt-redirect=redirect-server> Content-Length: 0 [Time: 13:1:37]

26d:13h:1m:37s ( sip_stack)(116219 ) UdpRtxMngr::Remove 1 INVITE [Time: 13:1:37]

26d:13h:1m:37s ( lgr_flow)(116220 ) | |(SIPTU#28)302 State:Inviting(1020791712266200913137@10.110.31.251) [Time: 13:1:37]

26d:13h:1m:37s ( lgr_flow)(116221 ) ---- Outgoing SIP Message to 172.20.1.20:5060 ---- [Time: 13:1:37]

26d:13h:1m:37s ACK sip:6888@MySIPDomain.com SIP/2.0 Via: SIP/2.0/UDP 10.110.31.251;branch=z9hG4bKac1020794782 Max-Forwards: 70 From: "AudioCodes 2" <sip:6489@MySIPDomain.com>;tag=1c1020792010 To: <sip:6888@MySIPDomain.com>;tag=31942 Call-ID: 1020791712266200913137@10.110.31.251 CSeq: 1 ACK Contact: <sip:5404-LR-MP124@10.110.31.251> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.40A.015.004 Content-Length: 0 [Time: 13:1:37]

26d:13h:1m:37s ( sip_stack)(116223 ) SIPCall(#28) changes state from Inviting to Disconnected [Time: 13:1:37]

26d:13h:1m:37s ( lgr_flow)(116224 ) | | | #14:SIP_DISCONNECT_EV(1020791712266200913137@10.110.31.251) [Time: 13:1:37]

26d:13h:1m:37s ( lgr_stk_mngr)(116225 ) Resource StackSession <#13> Allocated [Time: 13:1:37]

26d:13h:1m:37s ( lgr_stk_ses)(116226 ) <SESSION #14> SendToCall - event: INITIATE_REPLACING_CALL m_Call = 65115544 [Time: 13:1:37]

26d:13h:1m:37s ( lgr_flow)(116227 ) | | #14:INITIATE_REPLACING_CALL:(1020791712266200913137@10.110.31.251) [Time: 13:1:37]

26d:13h:1m:37s ( lgr_callf)(116229 ) new Call created - #13 [Time: 13:1:37]

26d:13h:1m:38s Device: 1. HO:1 [Code:3a002] [Time: 13:1:38]

26d:13h:1m:38s Device: 2. HO:1 [Code:3a002] [Time: 13:1:38]

26d:13h:1m:38s Device: 3. HO:1 [Code:3a002] [Time: 13:1:38]

26d:13h:1m:38s Device: 4. HO:1 [Code:3a002] [Time: 13:1:38]

26d:13h:1m:38s Device: 5. HO:1 [Code:3a002] [Time: 13:1:38]

26d:13h:1m:41s ( lgr_flow)(116296 ) ---- Incoming SIP Message from 172.20.103.253:5060 ---- [Time: 13:1:41]

26d:13h:1m:41s SIP/2.0 200 OK From: "AudioCodes 2"<sip:6489@MySIPDomain.com>;tag=1c1020907650 To: <sip:6888;phone-context=ChevyChase.Corporate@MySIPDomain.com>;tag=1f47b538-fd6714ac-13c4-40030-fb49e-fe3027c-fb49e Call-ID: 1020907337266200913137@10.110.31.251 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.110.31.251;branch=z9hG4bKac1020910803 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31 P-Asserted-Identity: "My CPND Name"<sip:6888;phone-context=ChevyChase.Corporate@MySIPDomain.com;user=phone> Privacy: none Contact: <sip:1036888;phone-context=ChevyChase.Corporate@MySIPDomain.com:5060;maddr=172.20.103.253;transport=udp;user=phone> Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 229 v=0 o=- 4003 1 IN IP4 172.20.103.253 s=- t=0 0 m=audio 5200 RTP/AVP 0 101 111 c=IN IP4 172.20.1.208 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv [Time: 13:1:41]

26d:13h:1m:41s ( lgr_flow)(116298 ) | |(SIPTU#27)200 OK State:Proceeding(1020907337266200913137@10.110.31.251) [Time: 13:1:41]

26d:13h:1m:41s ( sip_stack)(116299 ) SIPCall(#27) changes state from Proceeding to RemoteAccepted [Time: 13:1:41]

26d:13h:1m:41s ( lgr_flow)(116300 ) | | | #13:SIP_CONNECT_EV(1020907337266200913137@10.110.31.251) [Time: 13:1:41]

26d:13h:1m:41s ( sip_stack)(116301 ) SIPSDPSession#13 - Changing state from SIP_MEDIA_OFFERING to SIP_MEDIA_COMPLETED [Time: 13:1:41]

26d:13h:1m:41s ( lgr_flow)(116302 ) | |(SIPTU#27)ACK_REQ State:RemoteAccepted(1020907337266200913137@10.110.31.251) [Time: 13:1:41]

26d:13h:1m:41s ( sip_stack)(116303 ) SIPCall(#27) changes state from RemoteAccepted to Connected [Time: 13:1:41]

26d:13h:1m:41s ( lgr_flow)(116304 ) ---- Outgoing SIP Message to 172.20.103.253:5060 ---- [Time: 13:1:41]

26d:13h:1m:41s ( lgr_stk_ses)(116306 ) DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 13:1:41]

26d:13h:1m:42s ( lgr_flow)(116334 ) | | TransactionUserMngr::ReturnSIPCall - #28 [Time: 13:1:42]

26d:13h:1m:42s ( sip_stack)(116335 ) SIPCall(#28) changes state from Disconnected to Idle [Time: 13:1:42]

26d:13h:1m:43s Device: 3. HO:1 [Code:3a002] [Time: 13:1:43]

26d:13h:1m:43s Device: 5. HO:1 [Code:3a002] [Time: 13:1:43]

26d:13h:1m:43s ( sip_stack)(116336 ) new AcSIPDialogAPI created - #37 [Time: 13:1:43]

26d:13h:1m:43s ( lgr_flow)(116337 ) | | new GetNewIndTransaction created - #37 [Time: 13:1:43]

26d:13h:1m:43s ( lgr_flow)(116338 ) | |(SIPTU#37)DIALOG_INIT_REQ State:DialogIdle() [Time: 13:1:43]

26d:13h:1m:43s ( sip_stack)(116339 ) SIPDialog(#37) changes state from DialogIdle to DialogInitiated [Time: 13:1:43]

26d:13h:1m:43s ( lgr_flow)(116340 ) | |(SIPTU#37)GENERAL_REQ State:DialogInitiated(1331875659256200995056@10.110.31.251) [Time: 13:1:43]

26d:13h:1m:43s ( lgr_flow)(116341 ) ---- Outgoing SIP Message to 172.20.1.20:5060 ---- [Time: 13:1:43]

Anyway, sorry for the length of the post, but maybe a SIP trace log of a successful call will help with your troubleshooting.

 

[LuckyThirteen]

Ok, so your CDP and UDP domain names are "ChevyChase" and "Corporate"? If so, your SIP messages look very similar to mine. My CDP and UDP domain names are "cdp" and "udp".

I do not have a trace of a successful call because we haven't completed one yet, but I do have a trace of what we are getting currently. Here it is:

09:54:34 SIP.STACK MSG Tx: TCP src=10.19.253.6:5060 dst=172.22.231.245:5060
09:54:34 SIP.STACK MSG INVITE sip:6098@adtransip.com:5060 SIP/2.0
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 1 INVITE
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bcd9-761566f2
09:54:34 SIP.STACK MSG Max-Forwards: 70
09:54:34 SIP.STACK MSG Supported: 100rel,replaces
09:54:34 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:54:34 SIP.STACK MSG User-Agent: ADTRAN_NetVanta_7100/A2.03.13.E
09:54:34 SIP.STACK MSG Contact: <sip:3029@10.19.253.6:5060;transport=TCP>
09:54:34 SIP.STACK MSG Content-Type: application/SDP
09:54:34 SIP.STACK MSG Content-Length: 291
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG v=0
09:54:34 SIP.STACK MSG o=- 1246028063 1246028063 IN IP4 10.19.216.71
09:54:34 SIP.STACK MSG s=Polycom IP Phone
09:54:34 SIP.STACK MSG c=IN IP4 10.19.216.71
09:54:34 SIP.STACK MSG t=0 0
09:54:34 SIP.STACK MSG a=sendrecv
09:54:34 SIP.STACK MSG m=audio 50058 RTP/AVP 0 18 101
09:54:34 SIP.STACK MSG a=silenceSuppff - - - -
09:54:34 SIP.STACK MSG a=rtpmap:0 PCMU/8000
09:54:34 SIP.STACK MSG a=rtpmap:18 G729/8000
09:54:34 SIP.STACK MSG a=fmtp:18 annexb=no
09:54:34 SIP.STACK MSG a=rtpmap:101 telephone-event/8000
09:54:34 SIP.STACK MSG a=fmtp:101 0-15
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG Rx: TCP src=172.22.231.245:5060 dst=10.19.253.6:5060
09:54:34 SIP.STACK MSG SIP/2.0 302 Moved Temporarily
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bcd9-761566f2;received=10.19.253.6
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>;tag=20974
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 1 INVITE
09:54:34 SIP.STACK MSG Contact: <sip:6098;phone-context=cdp.udp@adtransip.com:5060;maddr=172.22.231.244;transport=tcp;x-nt-redirect=redirect-server>
09:54:34 SIP.STACK MSG Content-Length: 0
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG Tx: TCP src=10.19.253.6:5060 dst=172.22.231.245:5060
09:54:34 SIP.STACK MSG ACK sip:6098@adtransip.com:5060;transport=TCP SIP/2.0
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>;tag=20974
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 1 ACK
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bcd9-761566f2
09:54:34 SIP.STACK MSG Max-Forwards: 70
09:54:34 SIP.STACK MSG Supported: 100rel,replaces
09:54:34 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:54:34 SIP.STACK MSG User-Agent: ADTRAN_NetVanta_7100/A2.03.13.E
09:54:34 SIP.STACK MSG Contact: <sip:3029@10.19.253.6:5060;transport=TCP>
09:54:34 SIP.STACK MSG Content-Length: 0
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG Tx: TCP src=10.19.253.6:5060 dst=172.22.231.245:5060
09:54:34 SIP.STACK MSG INVITE sip:6098;phone-context=cdp.udp@adtransip.com:5060;maddr=172.22.231.244;transport=tcp;x-nt-redirect=redirect-server SIP/2.0
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 2 INVITE
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bd05-3462f138
09:54:34 SIP.STACK MSG Max-Forwards: 70
09:54:34 SIP.STACK MSG Supported: 100rel,replaces
09:54:34 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:54:34 SIP.STACK MSG User-Agent: ADTRAN_NetVanta_7100/A2.03.13.E
09:54:34 SIP.STACK MSG Contact: <sip:3029@10.19.253.6:5060;transport=TCP>
09:54:34 SIP.STACK MSG Content-Type: application/SDP
09:54:34 SIP.STACK MSG Content-Length: 291
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG v=0
09:54:34 SIP.STACK MSG o=- 1246028063 1246028063 IN IP4 10.19.216.71
09:54:34 SIP.STACK MSG s=Polycom IP Phone
09:54:34 SIP.STACK MSG c=IN IP4 10.19.216.71
09:54:34 SIP.STACK MSG t=0 0
09:54:34 SIP.STACK MSG a=sendrecv
09:54:34 SIP.STACK MSG m=audio 50058 RTP/AVP 0 18 101
09:54:34 SIP.STACK MSG a=silenceSuppff - - - -
09:54:34 SIP.STACK MSG a=rtpmap:0 PCMU/8000
09:54:34 SIP.STACK MSG a=rtpmap:18 G729/8000
09:54:34 SIP.STACK MSG a=fmtp:18 annexb=no
09:54:34 SIP.STACK MSG a=rtpmap:101 telephone-event/8000
09:54:34 SIP.STACK MSG a=fmtp:101 0-15
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG Rx: TCP src=172.22.231.245:5060 dst=10.19.253.6:5060
09:54:34 SIP.STACK MSG SIP/2.0 302 Moved Temporarily
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bd05-3462f138;received=10.19.253.6
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>;tag=16445
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 2 INVITE
09:54:34 SIP.STACK MSG Contact: <sip:6098;phone-context=cdp.udp@adtransip.com:5060;maddr=172.22.231.244;transport=tcp;x-nt-redirect=redirect-server>
09:54:34 SIP.STACK MSG Content-Length: 0
09:54:34 SIP.STACK MSG
09:54:34 SIP.STACK MSG Tx: TCP src=10.19.253.6:5060 dst=172.22.231.245:5060
09:54:34 SIP.STACK MSG ACK sip:6098;phone-context=cdp.udp@adtransip.com:5060;maddr=172.22.231.244;transport=tcp;x-nt-redirect=redirect-server SIP/2.0
09:54:34 SIP.STACK MSG From: "Daniel Ogles" <sip:3029@adtransip.com:5060;transport=TCP>;tag=4d748a8-0-13c4-24fce-6154861d-24fce
09:54:34 SIP.STACK MSG To: <sip:6098@adtransip.com:5060>;tag=16445
09:54:34 SIP.STACK MSG Call-ID: 4db6d28-0-13c4-24fce-669f1131-24fce@adtransip.com
09:54:34 SIP.STACK MSG CSeq: 2 ACK
09:54:34 SIP.STACK MSG Via: SIP/2.0/TCP 10.19.253.6:5060;branch=z9hG4bK-24fce-907bd05-3462f138
09:54:34 SIP.STACK MSG Max-Forwards: 70
09:54:34 SIP.STACK MSG Supported: 100rel,replaces
09:54:34 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:54:34 SIP.STACK MSG User-Agent: ADTRAN_NetVanta_7100/A2.03.13.E
09:54:34 SIP.STACK MSG Contact: <sip:3029@10.19.253.6:5060;transport=TCP>
09:54:34 SIP.STACK MSG Content-Length: 0

The problem may well be in the Adtran, but I have no way of knowing for sure or proving it. It seems like I'm the first one to ever try to network these two boxes, so there's just not any info out there that I can find.

 

[allenmac]

It looks like you are almost there.

I was having a problem several months ago with SIP calls not flowing properly between networked CS1000E's, and I worked with Nortel ETAS for a couple of days on it.

The solution was to change my transport protocol from TCP to UDP. apparently, there was a lot of TCP message fragmentation going on in my network, and the SIP control messages were getting mixed up - a partial message was recieved, then another message, then a retransmission of the partial message, etc. Anyway, this may help.

 

[LuckyThirteen]

Thanks allenmac, I will give that a shot.