Re: [Linphone-developers] linphonec 3.4.0 fails to initiate SIP calls

[Simon Morlat]

Ok, thank you for the notification, I feel better now !

Simon

Le samedi 12 février 2011 à 12:14 -0500, Pedro Sanchez a écrit :
> On 02/11/2011 09:25 PM, Pedro Sanchez wrote:
> > Hello,
> >
> > I downloaded linphonec 3.4.0 and cross-compiled it for ARM. I had to
> > assume OSIP_SUCCESS == 0 for the compilation to succeed according to
> > this e-mail:
> >
> > http://lists.nongnu.org/archive/html/linphone-developers/2011-02/msg00061.html
> >
> >
> > Linphone successfully receives calls, audio and everything else works,
> > but I can't get it to initiate calls. This is what I get:
> >
> > linphonec> call sip:address@hidden
> > Establishing call id to <sip:address@hidden>, assigned id 1
> > linphonec> Contacting <sip:address@hidden>
> > linphonec> Call 1 to <sip:address@hidden> in progress.
> > Remote ringing.
> > linphonec> Call 1 to <sip:address@hidden> ringing.
> > Warning: Remote end <sip:address@hidden> seems to have
> > disconnected, the call is going to be closed.
> > linphonec> Call ended
> > linphonec> Call 1 with <sip:address@hidden> ended.
> > ortp-error-Could not remove the call from the list !!!
> > Call 1 with <sip:address@hidden> error.
> >
> > A SIP trace shows the INVITE from linphone followed by the TRYING and
> > RINGING from the remote end. This is followed almost immediately by a
> > CANCEL from linphonec.
> >
> > I must say that once in a blue moon it works but I can't tell what
> > exactly is necessary for this to work. For sure it doesn't work on the
> > very first call attempt. And as I keep trying to call I notice that the
> > call id numbers are always increasing, is this expected?
> >
> > Also, when I exit I get this:
> >
> > linphonec> quit
> > Terminating...
> > *** glibc detected *** linphonec: munmap_chunk(): invalid pointer:
> > 0x00038bc0 ***
> > Aborted
> >
> > I was running linphonec 3.3.2 before on the same host, calling the same
> > peer, and linphonec never failed. Any pointers on what to do would be
> > appreciated.
> >
> > Thank you,
> >
> >
> 
> The problem was a patch that I forgot to mention which aims at making 
> the RTP timeout event that ends a call on media inactivity work from the 
> beginning of the SIP negotiation and not from the moment the first RTP 
> packet is received.
> 
> I was given a patch that was discussed somewhere else which indeed 
> implements the feature but unfortunately breaks the initiation of new 
> calls. I attach a modified patch that fixes the problem in case someone 
> else is interested.
> 
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