#402: Call transfer not working with asterisk and same lan
-----------------------------+-------------------------
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:8 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)
You are correct. The tcpdump command was too restrictive. Now it should be
fixed. I have attatched a new dump.
On this one behaviour is like this: I could put the call on hold few times
and last 2 times it did not recover from it. I can hear the "repeating"
sound after resuming from "hold".
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Ticket URL: <http://www.qutecom.org/ticket/402#comment:9>
QuteCom <http://trac.qutecom.org>
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