CISCO SIP NO RTP PACKETS

[BAYSHORE]

I have CCM 4.1 going through a 2811 which has a SIP trunk to 3CX. I have a Linksys IP phone and a 3CX softphone connected to 3cx

The TRUNK is regsitered and outbound calls ......from 3cx to Cisco Phones work well but when I call from CISCO to 3CX the phone will ring but when you answer it the CISCO goes busy and the phones on the 3CX side says connected.


WHen I run Wireshark on 3cx Server and CCM I see when you make a outbound call (3cx to CISCO) you can see the RTP packets form the phone to 3cx and then 3cx to CCM...But when you make an inbound call (CISCO to 3CX) you see the phone sending RTP stream directly to CCM but nothing in return.


IN 3CX outbound ( 3cx to CISCO) you see that it connects ext 4004 and trunk 10000 in sa call
but in bound (CISCO to 3CX) it tries to tie cisco phone 1010 and 3cx ext 4004 in a call


here are the logs

the good one
16:15:54.375 Call::Terminate [CM503008]: Call(5): Call is terminated 16:15:54.359 Call::RouteFailed [CM503015]: Call(5): Attempt to reach [sip:[email protected]] failed. Reason: Not Found 16:15:54.359 CallLeg:nFailure [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:192.168.1.15:5060 16:15:54.281 MediaServerReporting::SetRemoteParty [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 192.168.1.100:7088(7089) 16:15:54.281 CallCtrl:nSelectRouteReq [CM503004]: Call(5): Calling: VoIPline:1003@(Ln.10000@CISCO_CCM)@[Dev:sip:[email protected]:5060] 16:15:54.265 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]] 16:15:54.265 MediaServerReporting::SetRemoteParty [MS210000] C:5.1:Offer received. RTP connection: 192.168.1.30:16410(16411) 16:15:54.265 CallLeg::setRemoteSdp Remote SDP is set for legC:5.1 16:15:54.250 Extension:rintEndpointInfo [CM505001]: Ext.4004: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.100:5060] 16:15:54.250 CallCtrl:nIncomingCall [CM503001]: Call(5): Incoming call from Ext.4004 to [sip:[email protected]] 16:15:54.250 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK-7dd98b63
Max-Forwards: 70
Contact: "4004"[sip:[email protected]:5060]
To: [sip:[email protected]]
From: "4004"[sip:[email protected]];tag=465d49f7b1b1711do0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="4004",realm="3CXPhoneSystem",nonce="12923698554:6c3528d83052e46164c7610b1e961aea",uri="sip:[email protected]",algorithm=MD5,response="4aa562f4433630a50ff075a38669cb51"
Supported: replaces
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0



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The bad one


12:32:12.031 Call::Terminate [CM503008]: Call(34): Call is terminated
12:31:44.062 Call::Terminate [CM503008]: Call(34): Call is terminated
12:31:39.984 CallCtrl:nLegConnected [CM503007]: Call(34): Device joined: sip:[email protected]:5060
12:31:39.984 CallCtrl:nLegConnected [CM503007]: Call(34): Device joined: sip:[email protected]
12:31:39.968 MediaServerReporting::SetRemoteParty [MS210007] C:34.1:Answer provided. Connection(by pass mode): 192.168.1.30:16458(16459)
12:31:39.968 MediaServerReporting::SetRemoteParty [MS210001] C:34.2:Answer received. RTP connection: 192.168.1.30:16458(16459)
12:31:39.968 Extension:rintEndpointInfo [CM505001]: Ext.4004: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.100:5060]
12:31:39.968 CallLeg::setRemoteSdp Remote SDP is set for legC:34.2
12:31:39.968 CallCtrl:nAnsweredCall [CM503002]: Call(34): Alerting sip:[email protected]:5060
12:31:38.640 MediaServerReporting::SetRemoteParty [MS210006] C:34.2:Offer provided. Connection(by pass mode): 173.100.18.49:24728(24729)
12:31:38.625 CallCtrl:nSelectRouteReq [CM503004]: Call(34): Calling: Ext:Ext.4004@[Dev:sip:[email protected]:5060]
12:31:38.625 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
12:31:38.625 MediaServerReporting::SetRemoteParty [MS210000] C:34.1:Offer received. RTP connection: 173.100.18.49:24728(24729)
12:31:38.625 CallLeg::setRemoteSdp Remote SDP is set for legC:34.1
12:31:38.625 Extension:rintEndpointInfo [CM505001]: Ext.1003: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.1.100:5060]
12:31:38.625 CallCtrl:nIncomingCall [CM503001]: Call(34): Incoming call from Sip.1003 to [sip:[email protected]] 12:31:38.609 MediaServerReporting::STUN [MS101000] C:34.1: Requested STUN server "stun.3cx.com:3478" not found
12:31:36.359 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 173.100.18.49;branch=z9hG4bK2bf557ea
Max-Forwards: 70
Contact: [sip:[email protected]:5060;transport=tcp]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=16777375
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 180
Min-SE: 1800
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Date: Fri, 16 Jul 2010 21:03:10 GMT
Supported: timer
User-Agent: Cisco-CCM4.1
Allow-Events: telephone-event
Content-Length: 0
Remote-Party-ID: [sip:[email protected]];party=calling;screen=no;privacy=off

12:30:42.578 Call::Terminate [CM503008]: Call(33): Call is terminated 12:30:42.578 Call::Terminate [CM503008]: Call(33): Call is terminated 12:30:34.125 CallLeg:nConfirmed Session 2504 of leg C:33.1 is confirmed 12:30:34.015 CallCtrl:nLegConnected [CM503007]: Call(33): Device joined: sip:[email protected]:5060 12:30:34.015 CallCtrl:nLegConnected [CM503007]: Call(33): Device joined: sip:[email protected]:5060 12:30:34.015 MediaServerReporting::SetRemoteParty [MS210003] C:33.1:Answer provided. Connection(transcoding mode):192.168.1.100:7160(7161) 12:30:34.015 MediaServerReporting::SetRemoteParty [MS210001] C:33.2:Answer received. RTP connection: 173.100.18.49:24724(24725) 12:30:34.015 CallLeg::setRemoteSdp Remote SDP is set for legC:33.2




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thanks in advance

 

[leejor]

the good one
16:15:54.375 Call::Terminate [CM503008]: Call(5): Call is terminated 16:15:54.359 Call::RouteFailed [CM503015]: Call(5): Attempt to reach [sip:[email protected]] failed. Reason: Not Found 16:15:54.359 CallLeg:nFailure [CM503003]: Call(5): Call to sip:1003@192.168.1.15:5060 has failed; Cause: 404 Not Found; from IP:192.168.1.15:5060 16:15:54.281 MediaServerReporting::SetRemoteParty [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 192.168.1.100:7088(7089) 16:15:54.281 CallCtrl:nSelectRouteReq [CM503004]: Call(5): Calling: VoIPline:1003@(Ln.10000@CISCO_CCM)@[Dev:sip:[email protected]:5060] 16:15:54.265 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]] 16:15:54.265 MediaServerReporting::SetRemoteParty [MS210000] C:5.1:Offer received. RTP connection: 192.168.1.30:16410(16411) 16:15:54.265 CallLeg::setRemoteSdp Remote SDP is set for legC:5.1 16:15:54.250 Extension:rintEndpointInfo [CM505001]: Ext.4004: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.100:5060] 16:15:54.250 CallCtrl:nIncomingCall [CM503001]: Call(5): Incoming call from Ext.4004 to [sip:[email protected]] 16:15:54.250 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK-7dd98b63
Max-Forwards: 70
Contact: "4004"[sip:[email protected]:5060]
To: [sip:[email protected]]
From: "4004"[sip:[email protected]];tag=465d49f7b1b1711do0
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="4004",realm="3CXPhoneSystem",nonce="12923698554:6c3528d83052e46164c7610b1e961aea",uri="sip:[email protected]",algorithm=MD5,response="4aa562f4433630a50ff075a38669cb51"
Supported: replaces
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0



**************************************************************************************************************
**************************************************************************************************************

The bad one


12:32:12.031 Call::Terminate [CM503008]: Call(34): Call is terminated
12:31:44.062 Call::Terminate [CM503008]: Call(34): Call is terminated
12:31:39.984 CallCtrl:nLegConnected [CM503007]: Call(34): Device joined: sip:[email protected]:5060
12:31:39.984 CallCtrl:nLegConnected [CM503007]: Call(34): Device joined: sip:[email protected]
12:31:39.968 MediaServerReporting::SetRemoteParty [MS210007] C:34.1:Answer provided. Connection(by pass mode): 192.168.1.30:16458(16459)
12:31:39.968 MediaServerReporting::SetRemoteParty [MS210001] C:34.2:Answer received. RTP connection: 192.168.1.30:16458(16459)
12:31:39.968 Extension:rintEndpointInfo [CM505001]: Ext.4004: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA941-5.1.8] Transport: [sip:192.168.1.100:5060]
12:31:39.968 CallLeg::setRemoteSdp Remote SDP is set for legC:34.2
12:31:39.968 CallCtrl:nAnsweredCall [CM503002]: Call(34): Alerting sip:[email protected]:5060
12:31:38.640 MediaServerReporting::SetRemoteParty [MS210006] C:34.2:Offer provided. Connection(by pass mode): 173.100.18.49:24728(24729)
12:31:38.625 CallCtrl:nSelectRouteReq [CM503004]: Call(34): Calling: Ext:Ext.4004@[Dev:sip:[email protected]:5060]
12:31:38.625 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
12:31:38.625 MediaServerReporting::SetRemoteParty [MS210000] C:34.1:Offer received. RTP connection: 173.100.18.49:24728(24729)
12:31:38.625 CallLeg::setRemoteSdp Remote SDP is set for legC:34.1
12:31:38.625 Extension:rintEndpointInfo [CM505001]: Ext.1003: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.1.100:5060]
12:31:38.625 CallCtrl:nIncomingCall [CM503001]: Call(34): Incoming call from Sip.1003 to [sip:[email protected]] 12:31:38.609 MediaServerReporting::STUN [MS101000] C:34.1: Requested STUN server "stun.3cx.com:3478" not found
12:31:36.359 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 173.100.18.49;branch=z9hG4bK2bf557ea
Max-Forwards: 70
Contact: [sip:[email protected]:5060;transport=tcp]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=16777375
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 180
Min-SE: 1800
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Date: Fri, 16 Jul 2010 21:03:10 GMT
Supported: timer
User-Agent: Cisco-CCM4.1
Allow-Events: telephone-event
Content-Length: 0
Remote-Party-ID: [sip:[email protected]];party=calling;screen=no;privacy=off

12:30:42.578 Call::Terminate [CM503008]: Call(33): Call is terminated 12:30:42.578 Call::Terminate [CM503008]: Call(33): Call is terminated 12:30:34.125 CallLeg:nConfirmed Session 2504 of leg C:33.1 is confirmed 12:30:34.015 CallCtrl:nLegConnected [CM503007]: Call(33): Device joined: sip:[email protected]:5060 12:30:34.015 CallCtrl:nLegConnected [CM503007]: Call(33): Device joined: sip:[email protected]:5060 12:30:34.015 MediaServerReporting::SetRemoteParty [MS210003] C:33.1:Answer provided. Connection(transcoding mode):192.168.1.100:7160(7161) 12:30:34.015 MediaServerReporting::SetRemoteParty [MS210001] C:33.2:Answer received. RTP connection: 173.100.18.49:24724(24725) 12:30:34.015 CallLeg::setRemoteSdp Remote SDP is set for legC:33.2

I suspect that some packets are not able to get through your firewall/router which is also causing the STUN server failure message in the second half of your post.

In the first part where 1003 (an external extension) is not found, that also, is probably the result of ports being blocked somewhere, firewall or router. Run the 3CX firewall checker and see if it provides any clues. be sure that you have opened up all necessary ports on the local router.

 

[BAYSHORE]

Thank you leejor

I will run the firewall checker and look into any ports that are blocked.

I am not running any firewall on my 2811. Are these ports block automatically? Do I need to open these specfic ports up? It seems odd that the RTP stream is working when you call from the linksys phone (linksys phone -> 3CX -> 2811 -> CCM -> CISCO Phones)
but when I try the other way ( CISCO Phones -> CCM -> 2811 -> 3CX -> linksys phone) the RTP traffice seems blocked.

Another thing that confuses me is when I run Wireshark and make the call (linksys phone -> 3CX -> 2811 -> CCM -> CISCO Phones) you can see the RTP stream flow through the SIP TRunk but when I make the call ( CISCO Phones -> CCM -> 2811 -> 3CX -> linksys phone) it looks the the LINKSYS phones are sending RTP packets directly to the CCM ???


Thanks again for your help

 

[BAYSHORE]

I found two issues that was caused the lost RTP packets.

The first I spoted in the 3csx log when the call started the CISCO phone sent the call set up info using TCP. The calls that worked the liksys sent this using UDP once i changed the CISCO phone to UDP the calls would go through but still would loose the RTP pcakets along the way.

The second thing I noticed from the 3CX logs was the path calls that went through linksys-3CX-2811-CCM-7920 where going through the SIP trunk. In the logs it showed a transcoding mode in the connection string. In the calls that did not go through the 7920-CCM-2811-3CX-linksys had a by pass mode (even though they where on seperate IP subnets). I forced these calls through the SIP trunk by checking on the trunk and on the Linksys extension in 3cx the PBX delivers audio check box.


Now all calls go through the SIP trunk and I am not loosing any RTP packtes


Thank you all for your help

 

[Administrador]

Hello,
I have made a trunk with Cisco Call Manager, and everything works fine, except for one thing,
When I call from a Cisco Cal Manager (CCM) extension, to a 3CX extension and the 3CX extension doesn't answer it should redirect the call to a voicemail, but it doesn't I have just the busy tone, Could "CISCO SIP NO RTP PACKETS" case be related to what happens to me?

Thanks,


Andrés M.