SIP packets dropped ?

[michaelbird75]

I have a 3CX connected via private LAN to a SIP server. Inbound calls work. Outbound calls are not working right now.

Outbound call from 3CX to cell captured on wireshark from local interface of 3CX server:
T38 sends invite to3CX
3CX sends invite to SIP
SIP sends 100 trying
3CX sends another invite to SIP
Sip sends a 183 progress
3CX sends invite
SIP begins RTP stream
3CX sends invite

When i check the 3CX server interface I see the SIP requests timing out as if no packets have been received from the SIP server.

Any thoughts?

Verbose below:

Wireshark:
|Time | 172.18.132.1 | 10.16.0.245 |
| | | 10.16.0.246 |
|41.901752000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|41.904791000| 100 Trying| | |SIP Status
| |(15706) <------------------ (5060) | |
|42.401850000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|42.404417000| 100 Trying| | |SIP Status
| |(15706) <------------------ (5060) | |
|43.401841000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|43.404126000| 100 Trying| | |SIP Status
| |(15706) <------------------ (5060) | |
|45.476999000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|45.478860000| 100 Trying| | |SIP Status
| |(15706) <------------------ (5060) | |
|47.785779000| 183 Session Progress | |SIP Status
| |(15706) <------------------ (5060) | |
|47.909154000| RTP (g711U) | |RTP Num packets:1056 Duration:21.117s SSRC:0x12D86402
| |(7010) <-------------------------------------- (20224) |
|49.477017000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|49.479641000| 183 Session Progress | |SIP Status
| |(15706) <------------------ (5060) | |
|52.788045000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|53.285824000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|54.285856000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|56.285464000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|57.506147000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|57.508430000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|60.284928000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|64.285343000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|68.284643000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |
|72.284873000| BYE | | |SIP Request
| |(5060) <------------------ (15725) | |
|72.385245000| 481 Call/Transaction | |SIP Status
| |(5060) ------------------> (5060) | |
|73.521309000| INVITE SDP (g711U g7 | |SIP From: "14444356475"<sip:[email protected]:5060 To:<sip:[email protected]:5060
| |(5060) ------------------> (5060) | |
|73.524065000| 200 OK SDP (g711U te | |SIP Status
| |(15706) <------------------ (5060) | |


3Cx side:
19-Jun-2015 16:44:45.412 L:4.1[Extn]: Terminating targets, reason:
19-Jun-2015 16:44:45.412 Leg L:4.1[Extn] is terminated: Cause: BYE from PBX
19-Jun-2015 16:44:45.412 Terminated from "Jacklyn"<sip:[email protected]>;tag=1601723566 to <sip:[email protected]>;tag=8420fb4f; reason: Rejected
19-Jun-2015 16:44:45.412 L:4.1[Extn] Sending: OnSendResp Send 480/INVITE from 0.0.0.0:0 tid=1924022360 [email protected]:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 172.18.132.13:5062;branch=z9hG4bK1924022360
To: <sip:[email protected]>;tag=8420fb4f
From: "Jacklyn"<sip:[email protected]>;tag=1601723566
Call-ID: [email protected]
CSeq: 2 INVITE
Warning: 499 3CX-PC "No answer"
Content-Length: 0
19-Jun-2015 16:44:45.412 SendMsg from <sip:[email protected]>;tag=8420fb4f to "Jacklyn"<sip:[email protected]>;tag=1601723566
19-Jun-2015 16:44:45.362 ~Target=VoIPline:2354455@(Ln.10000@wtf)
19-Jun-2015 16:44:45.362 Call(C:4) is terminated
19-Jun-2015 16:44:45.362 [CM503020]: Call(C:4): Normal call termination. Call originator: Extn:6700. Reason: No answer
19-Jun-2015 16:44:45.362 [CM503016]: Call(C:4): Attempt to reach <sip:[email protected]:5060> from Extn:6700 has failed. Reason: No Answer
19-Jun-2015 16:44:45.361 L:4.2[Line:10000>>2354455]: Terminating targets, reason: SIP ;cause=408 ;text="Request Timeout"
19-Jun-2015 16:44:45.360 [Flow] Current call diversion path:[]
19-Jun-2015 16:44:36.216 Currently active calls - 1: [4]
19-Jun-2015 16:44:13.250 [CM503025]: Call(C:4): Calling T:Line:10000>>2354455@[Dev:sip:[email protected]:5060] for L:4.1[Extn]
19-Jun-2015 16:44:13.249 Route to L:4.2[Line:10000>>2354455] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=756d8154373aae29 Call-ID=NGFmMTAzNTg0ZDk5MTU4NTA2MDQ4NDM3YzdiOWQxY2Q.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-756d8154373aae29-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "14425546457"<sip:[email protected]:5060>;tag=541a4225
Call-ID: NGFmMTAzNTg0ZDk5MTU4NTA2MDQ4NDM3YzdiOWQxY2Q.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 253
Remote-Party-ID: "14425546457"<sip:[email protected]:5060>;party=calling

v=0
o=3cxPS 49408901120 474241564673 IN IP4 172.18.132.1
s=3cxPS Audio call
c=IN IP4 172.18.132.1
t=0 0
m=audio 7014 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
19-Jun-2015 16:44:13.243 Outbound URI is used: sip:[email protected]:5060
19-Jun-2015 16:44:13.243 SLA is globally disabled
19-Jun-2015 16:44:13.243 Added leg L:4.2[Line:10000>>2354455]
19-Jun-2015 16:44:13.198 [Flow] Call(C:4): making call from L:4.1[Extn] to T:Line:10000>>2354455@[Dev:sip:[email protected]:5060]
19-Jun-2015 16:44:13.198 [CM503027]: Call(C:4): From: Extn:6700 ("Jacklyn" <sip:[email protected]:5060>) to T:Line:10000>>2354455@[Dev:sip:[email protected]:5060]
19-Jun-2015 16:44:13.198 [CM503004]: Call(C:4): Route 1: from L:4.1[Extn] to T:Line:10000>>2354455@[Dev:sip:[email protected]:5060]
19-Jun-2015 16:44:13.198 Line limit check: Current # of calls for line Lc:10000(@wtf[<sip:[email protected]:5060>]) is 0; limit is 12
19-Jun-2015 16:44:13.198 Call(C:4): Call from Extn:6700 to 92354455 matches outbound rule 'Rule for wtf'
19-Jun-2015 16:44:13.198 [Flow] Call(C:4): has built target endpoint: Out#:>>Rule{Rule for wtf}>>92354455 for call from L:4.1[Extn]
19-Jun-2015 16:44:13.198 [Flow] Target endpoint for 92354455 is Out#:>>Rule{Rule for wtf}>>92354455
19-Jun-2015 16:44:13.198 Selected prefix: 9
19-Jun-2015 16:44:13.198 Looking for outbound rule: dialed = [92354455], processed: [92354455]; from-ext:
19-Jun-2015 16:44:13.198 [Flow] Building target endpoint to 92354455 from "Jacklyn" <sip:[email protected]:5060>
19-Jun-2015 16:44:13.198 [CM503010]: Call(C:4): Making route(s) from Extn:6700 to <sip:[email protected]:5060>
19-Jun-2015 16:44:13.198 Remote SDP is set for leg L:4.1[Extn]
19-Jun-2015 16:44:13.198 OnOffer from "Jacklyn"<sip:[email protected]>;tag=1601723566
19-Jun-2015 16:44:13.198 [CM505001]: Endpoint Extn:6700: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Yealink SIP-T48G 35.72.0.225] PBX contact: [sip:[email protected]:5060]
19-Jun-2015 16:44:13.197 Inbound DID: ''; Phonebook Name: ''
19-Jun-2015 16:44:13.197 [CM500002]: Call(C:4): Info on incoming INVITE from Extn:6700:
Invite-IN Recv Req INVITE from 172.18.132.13:5062 tid=1924022360 [email protected]:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.18.132.13:5062;branch=z9hG4bK1924022360
Max-Forwards: 70
Contact: <sip:[email protected]:5062>
To: <sip:[email protected]>
From: "Jacklyn"<sip:[email protected]>;tag=1601723566
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Content-Type: application/sdp
Proxy-Authorization: Digest username="6700",realm="3CXPhoneSystem",nonce="414d535c0b951e2d61:2c87dfec7162863931a8140b3c83d0e0",uri="sip:[email protected]",response="9fc8c95b712a5c239876f90d396bfe85",algorithm=MD5
Supported: replaces
User-Agent: Yealink SIP-T48G 35.72.0.225
Allow-Events: talk, hold, conference, refer, check-sync
Content-Length: 308

v=0
o=- 20003 20003 IN IP4 172.18.132.13
s=SDP data
c=IN IP4 172.18.132.13
t=0 0
m=audio 11786 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
19-Jun-2015 16:44:13.197 [CM503001]: Call(C:4): Incoming call from Extn:6700 to <sip:[email protected]:5060>
19-Jun-2015 16:44:13.193 Outbound URI is used: sip:[email protected]:5062
19-Jun-2015 16:44:13.193 IncomingCall: C:4 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
19-Jun-2015 16:44:13.193 Added leg L:C:4.1[No endpoint yet]
19-Jun-2015 16:44:13.193 UasSession 716 started
19-Jun-2015 16:44:13.193 Call from "Jacklyn"<sip:[email protected]>;tag=1601723566 to <sip:[email protected]>;tag=8420fb4f
19-Jun-2015 16:44:04.214 Currently active calls [none]
19-Jun-2015 16:43:34.212 Currently active calls [none]
19-Jun-2015 16:43:02.211 Currently active calls [none]
19-Jun-2015 16:42:30.209 Currently active calls [none]
19-Jun-2015 16:41:58.207 Currently active calls [none]
19-Jun-2015 16:41:26.205 Currently active calls [none]
19-Jun-2015 16:40:54.203 Currently active calls [none]

 

[leejor]

Did this problem "just start", or has it been this way from the beginning. In other words, did something change on the network set-up/settings, that may be causing this?

Have you run the 3CX Firewall Checker with success?

Generally, when it's an incoming problem , with outgoing OK, then it is a routing issue, or perhaps a setting in the router. Listing what make/model router you are using may assist other in providing the (perhaps) required settings to have SIP work correctly.