[sip-comm-dev] SIP Communicator with Asterisk ?

asmouta asmouta at gmail.com
Tue Sep 25 13:00:18 CEST 2007


I'm sorry, I dont have the files here but I can ensure you that my
configuration is a very simple one. Have you tried with another kind of
client??

Ciao


On 9/25/07, Samir S <samir.bot at gmail.com> wrote:
>
> Thanks asmouta
> Could you send me the extensions.conf/sip.conf and asterisk.conf files,
> which you configured for that setup ?
>
> For me the issue, is that when I call from User1 to the User2 and the call
> is accepted, then after the ACKs, Asterisk sends a reINVITE to both the SC
> Endpoints,which are not replied to.
> Is there some option to disable this in Asterisk ? Hence, any help with
> the config files would be great.
>
> Thanks and Regards
> Samir
>
>
>
> On 9/25/07, asmouta <asmouta at gmail.com> wrote:
> >
> > I have already used SC with Asterisk, the only problem I faced was when
> > I used an old version of Asterisk which doesn't forward the SIP header field
> > max-forward. But since I installed the newest version all was ok.
> >
> > Ciao.
> >
> >
> >  On 9/25/07, Samir S <samir.bot at gmail.com> wrote:
> > >
> > > Does anyone have experience with SC + Asterisk ?
> > > It seems that the Asterisk sends a reINVITE (acting as a B2BUA) which
> > > is not supported by SC. ?
> > >
> > > Is there some branch, wherein SC supports reINVITEs ?
> > > Or the other way around (in case, someone on this mailing list knows),
> > > is it possible to make Asterisk work as a plain proxy to make it work with
> > > the SIP communicator
> > >
> > > Thanks and Regards
> > > Samir
> > >
> > >
> >
>
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