#402: Call transfer not working with asterisk and same lan
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Reporter: janno | Owner: vadim
Type: defect | Status: new
Priority: critical | Milestone: QuteCom 3.0
Component: 3rd party libs | Version: 2.2
Resolution: | Keywords:
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Comment (by janno):
Replying to [comment:6 vadim]:
> Janno,
>
> The dump is very strange it does not contains SIP traffic FROM
QuteCom...
>
> Can you make a wireshark capture on machine runing QuteCom?
What do you mean. Isn't that correct?:
717 23.361724 192.168.5.208 192.168.5.206 RTP 214
PT=ITU-T G.711 PCMU, SSRC=0x32FCA792, Seq=370, Time=59200
718 23.381728 192.168.5.208 192.168.5.206 RTP 214
PT=ITU-T G.711 PCMU, SSRC=0x32FCA792, Seq=371, Time=59360
719 23.401710 192.168.5.208 192.168.5.206 RTP 214
PT=ITU-T G.711 PCMU, SSRC=0x32FCA792, Seq=372, Time=59520
Seems like this "double data stream" occurs only then forwarding a call.
Nevertheless - it seems that things start to go wrong allready then I put
a call on hold. So we should start with that.
--
Ticket URL: <http://www.qutecom.org/ticket/402#comment:7>
QuteCom <http://trac.qutecom.org>
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