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Disconnection...

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GMBİL

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I'm using Vega 50 Europa 4FXO with 3CX latest version, i can manage to dial out but after 5-10 second the call looses audio, it seems still connected on both external and internal side but no audio...

Any ideas?

Below 3cx Logs,


10:34:26.763 [CM503008]: Call(78): Call is terminated

10:34:26.757 [CM503008]: Call(78): Call is terminated

10:33:51.026 Session 15113 of leg C:78.1 is confirmed

10:33:50.935 [CM503007]: Call(78): Device joined: sip:[email protected]:5060

10:33:50.920 [CM503007]: Call(78): Device joined: sip:[email protected]:5060

10:33:50.919 [CM503021]: Call(78): Call recording is started, audio file: C:\ProgramData\3CX\Data\Recordings\101\[Mehmet Ali Demirci]_101-905544021790_20090606103350(78).wav

10:33:50.915 [MS210003] C:78.1:Answer provided. Connection(transcoding mode):192.168.0.100:7354(7355)

10:33:50.914 [MS210001] C:78.2:Answer received. RTP connection: 192.168.0.110:10002(10003)

10:33:50.908 Remote SDP is set for legC:78.2

10:33:50.907 [CM505002]: Gateway:[VegaStream_50_4FXO] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VEGAEURO/13.02.08.6xS008] Transport: [sip:192.168.0.100:5060]

10:33:50.907 [CM503002]: Call(78): Alerting sip:[email protected]:5060

10:33:47.548 [CM503024]: Call(78): Calling PSTNline:05544021790@(Ln.10000@VegaStream_50_4FXO)@[Dev:sip:[email protected]:5060]

10:33:47.546 [MS210002] C:78.2:Offer provided. Connection(transcoding mode): 192.168.0.100:7356(7357)

10:33:47.537 [CM503004]: Call(78): Route 1: PSTNline:05544021790@(Ln.10000@VegaStream_50_4FXO)@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]

10:33:47.525 [MS210000] C:78.1:Offer received. RTP connection: 192.168.0.111:16406(16407)

10:33:47.524 [CM503010]: Making route(s) to <sip:[email protected]>

10:33:47.519 Remote SDP is set for legC:78.1

10:33:47.517 [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA921-4.1.10(b)] Transport: [sip:192.168.0.100:5060]

10:33:47.504 [CM503001]: Call(78): Incoming call from Ext.101 to <sip:[email protected]>

10:33:47.496 [CM500002]: Info on incoming INVITE:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.0.111:5060;branch=z9hG4bK-a3cb6cac

Max-Forwards: 70

Contact: "Mehmet Ali Demirci"<sip:[email protected]:5060>

To: <sip:[email protected]>

From: "Mehmet Ali Demirci"<sip:[email protected]>;tag=b13731bc44217b2o0

Call-ID: [email protected]

CSeq: 102 INVITE

Expires: 240

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c003aacdb79:bf9e7f12c5a0d2443fa9af4bd5276313",uri="sip:[email protected]",algorithm=MD5,response="eb34a45c4b6661e13fb9c31b112df60b"

User-Agent: Sipura/SPA921-4.1.10(b)

Content-Length: 0
 
It looks like you are recording the call, you might want to disable that while trouble shooting. You may also want to look at, and possibly change, what codecs you are using on each leg of the call. You may have a compatibility problem.
 
I disabled call recording, and selected other codecs to use the system but these even did not solve the problem...
 
When you now look at a log of the call do you still see (anywhere) "Offer provided. Connection(transcoding mode)" ?

I'm not familiar with the particular gateway that you are using. I'm wondering if it has the provision to send diagnostic logs to a syslog server. If that can be enabled (you can download a free syslog programme from Kiwi and run it on a computer on your network), it may give you a clue at to which end is causing the disconnect.
 
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