Dismiss Notice
We would like to remind you that we’re updating our login process for all 3CX forums whereby you will be able to login with the same credentials you use for the Partner or Customer Portal. Click here to read more.

Disconnection...

Discussion in '3CX Phone System - General' started by GMBİL, Jun 6, 2009.

Thread Status:
Not open for further replies.
  1. GMBİL

    Joined:
    Jun 4, 2009
    Messages:
    9
    Likes Received:
    0
    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:10000@192.168.0.110:5060

    10:33:50.920 [CM503007]: Call(78): Device joined: sip:101@192.168.0.111:5060

    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:10000@192.168.0.110:5060

    10:33:47.548 [CM503024]: Call(78): Calling PSTNline:05544021790@(Ln.10000@VegaStream_50_4FXO)@[Dev:sip:10000@192.168.0.110:5060]

    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:10000@192.168.0.110:5060, Dev:sip:10001@192.168.0.110:5060, Dev:sip:10002@192.168.0.110:5060, Dev:sip:10003@192.168.0.110:5060]

    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:905544021790@192.168.0.100>

    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:905544021790@192.168.0.100>

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

    INVITE sip:905544021790@192.168.0.100 SIP/2.0

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

    Max-Forwards: 70

    Contact: "Mehmet Ali Demirci"<sip:101@192.168.0.111:5060>

    To: <sip:905544021790@192.168.0.100>

    From: "Mehmet Ali Demirci"<sip:101@192.168.0.100>;tag=b13731bc44217b2o0

    Call-ID: 54a7f20f-dc14854e@192.168.0.111

    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:905544021790@192.168.0.100",algorithm=MD5,response="eb34a45c4b6661e13fb9c31b112df60b"

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

    Content-Length: 0
     
  2. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    11,073
    Likes Received:
    323
    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.
     
  3. GMBİL

    Joined:
    Jun 4, 2009
    Messages:
    9
    Likes Received:
    0
    I disabled call recording, and selected other codecs to use the system but these even did not solve the problem...
     
  4. leejor

    leejor Well-Known Member

    Joined:
    Jan 22, 2008
    Messages:
    11,073
    Likes Received:
    323
    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.
     
Thread Status:
Not open for further replies.