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


#1

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


#2

Hello all,
   
  Trying to record incoming message, I have synchronization / other problems.
   
  Please take a look at the recorded message: http://www.siptele.com/download/recordedMessage.mov
  and see if you can rescue me.
   
  Thanks,
   
  Michael


#3

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@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


#4

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@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@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
>
>


#5

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@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@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@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
> >
> >
>


#6

Thanks..It seems that Asterisk has some settings such as canreinvite=xxx for
controlling the media handling part..using reINVITEs
We can set it as "no" to stop Asterisk from sending reINVITE to re-target
the media directly between the clients.
Additionally, the directrtpsetup=yes setting tries to setup the media
peer-2-peer, without reINVITEs, ...but it does not seem to be working...:frowning:

From a SIP communicator perspective, there seems to be one more error.

It seems that the calling SC is stuck in some kind of loop, playing the
Alerting Ringback tone.
Even after accepting the call from the other end, it still continues to play
it, though the status in the CallParticipantPanel does say "Connected"...

I can understand that the RTP paths are with Asterisk now....and since
Asterisk is not sending the ReINVITEs..there *may* be some problem there
with regards to passing the media between the two SCs ...(though from a SC's
viewpoint, that should not be a problem)
.......but why does the Alerting Ringtone on the SIP communicator not stop
?

Thanks and Regards
Samir

···

On 9/25/07, asmouta <asmouta@gmail.com> wrote:

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@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@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@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
> > >
> > >
> >
>


#7

Hi,

I know that synchronization of audio in video is done by setting some timestamps.
But I think such problems will be more easy to track and fix after we move to FMJ.
For now tracking these kinf of problems in JMF is very painful.
I would only suggest looking at jmf mailing list as there is some mails about this,
but I think with no resolution.

damencho

mik wrote:

···

Hello all,
Trying to record incoming message, I have synchronization / other problems.
Please take a look at the recorded message: http://www.siptele.com/download/recordedMessage.mov
and see if you can rescue me.
Thanks,
Michael

---------------------------------------------------------------------
To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: dev-help@sip-communicator.dev.java.net