Re: [Linphone-users] linphonec 1.0.0pre8 / wrong phonenumber-message

[David Schumacher]

hello simon and aymeric,

thanks for your answers! i searched the code of the new 1.0.0pre9, 
starting at console/linphonec.c,
ending at coreapi/linphonecore.c at line 1040: int 
linphone_core_invite(LinphoneCore *lc, const char *url, 
LinphoneProxyConfig *proxy).
at this point i stopped to understand where the code ist going after 
that, like opening the audio-stream etc..
can you please give me a hint where to search for the condition-test in 
the sip-conversation, after that the audio-stream will be opened?

thanks in advance!
dave


Aymeric Moizard wrote:

> On Fri, 25 Feb 2005, Simon Morlat wrote:
>
> > Hello,
> >
> > It think the problem is that linphone does not understand properly 183
> > messages. Aymeric what do you think ? The 183 contains a sdp so linphone
> > should start the audio isn't it ?
>
>
> Yes. eXosip with the default negotiator is providing connection 
> information at this early stage. As you are doing the negotiation
> in linphone, you have to open stream in linphone when a provisionnal
> response for INVITE contains an SDP answer.
>
> Aymeric
>
> > Simon
> >
> > Le Jeudi 24 Février 2005 20:52, David Schumacher a écrit :
> >
> > > hello from germany,
> > >
> > > i'am uing linphonec 1.0.0pre8 with libosip2 2.2.0 with asterisk
> > > CVS-HEAD-02/03/05-22:32:08 as a voip <> isdn-gateway using chan_capi.
> > >
> > > i can call to the outside world and receive calls, but when i dial a 
> > > wrong
> > > number in linphonec (for example address@hidden), linphonec 
> > > keeps
> > > signalling the "remote phone is ringing"-tone for about 40 seconds. 
> > > if i
> > > use for example sjphone, i get the normal spoken message from the
> > > telephone-company (you have dialed a wrong number...).
> > >
> > > i tried ethereal to compare the sip-conversation of both phones.
> > > both phones started with:
> > >
> > > INVITE sip:address@hidden ...
> > > Status: 100 Trying
> > > Status: 183: Session Progress, with session description
> > >
> > > after that, there is some difference in the behaviour of the two 
> > > phones:
> > > with sjphone i can hear the message from the telephonecompany with 
> > > no more
> > > sip-conversation after Status: 183 till the message is over. 
> > > linphonec will
> > > keep "ringing" for the duration of the message, after that asterisk 
> > > sends a
> > > Status: 403 Forbidden.
> > >
> > > i already searched the sourcecode of libosip2, but i'am not a
> > > c-specialist...
> > >
> > > does anybody know a solution to make linphone transfer all audio as 
> > > soon as
> > > possible, including messages and tones from the phone-company, for 
> > > example
> > > like the (closed-source and unscriptable )-; ) sjphone?
> > >
> > > thanks in advance!
> > >
> > > dave
> > >
> > >
> > >
> > > _______________________________________________
> > > Linphone-users mailing list
> > > address@hidden
> > > http://lists.nongnu.org/mailman/listinfo/linphone-users