#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 vadim):
Replying to [comment:7 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
Sure, this is RTP traffic i'm talking about SIP traffic (reply to INVITE
request coming from Astreisk)
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Ticket URL: <http://www.qutecom.org/ticket/402#comment:8>
QuteCom <http://trac.qutecom.org>
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