[pjsip] prevent audible ring on client when called

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Yeah, I hear it on pjsua. I am investigating how to prevent this from
happening on asterisk. So far no luck. But thanks for pointing me in
the right direction.

On 10/25/07, Lafras Henning <lafras at xietel.com> wrote:
> Are you hearing the ring on PJSUA or on the other device?
>
> If you hear the ring on PJSUA it would be because the other device (not
> PJSUA) is sending a 180 to Asterisk, and Asterisk is producing a ring for
> that device.
> I do not know Asterisk well but I am sure you are able to set in the
> asterisk dial plan  for it not to produce the ring back.
>
> ----- Original Message -----
> From: "Danny Brown" <danbrwn@xxxxxxxxx>
> To: "pjsip embedded/DSP SIP discussion" <pjsip at lists.pjsip.org>
> Sent: Thursday, October 25, 2007 5:38 PM
> Subject: Re: [pjsip] prevent audible ring on client when called
>
>
> > Oh, maybe you are right. Do you think it is sending the ring in
> > response to the 100 sent out from pjsua. If so, anyway to bypass
> > sending the 100? And just send the 200.
> >
> > On 10/25/07, Lafras Henning <lafras at xietel.com> wrote:
> > > In this scenario once PJSUA answers, Asterisk would be streaming the
> audio
> > > to PJSUA,
> > > so if you hear a ring on PJSUA, it would be coming from Asterisk. Not
> so?
> > >
> > >
> > > ----- Original Message -----
> > > From: "Benny Prijono" <bennylp@xxxxxxxxx>
> > > To: "pjsip embedded/DSP SIP discussion" <pjsip at lists.pjsip.org>
> > > Sent: Thursday, October 25, 2007 5:04 AM
> > > Subject: Re: [pjsip] prevent audible ring on client when called
> > >
> > >
> > > > Danny Brown wrote:
> > > > > here is my config file passed to pjsua_vc6d.exe
> > > > > --id=sip:MAST100 at 192.168.13.100
> > > > > --registrar=sip:192.168.13.100
> > > > > --username=MAST100
> > > > > --realm=asterisk
> > > > > --password=1234
> > > > > --quality=3
> > > > > --auto-answer=200
> > > > > --ec-tail=600
> > > > >
> > > > > it registers, will accept calls but always rings once before
> > > > > answering. What am I missing? The way I am using this client is;
> > > > > I have an asterisk server running that is scripted to initiate a
> call
> > > > > from your sip client to another sip device. So, asterisk dials
> pjsua,
> > > > > once it answers it dials the other end and makes the connection. I
> do
> > > > > not want the caller (in this case pjsua) to actually ring ( at least
> > > > > not audibly). Thanks for your help. Dan
> > > >
> > > > Sorry I'm totally confused here. There is no such thing as audible
> > > > ringing in pjsua! The poor sample application just doesn't have this
> > > > feature!
> > > >
> > > >   -benny
> > > >
> > > >
> > > > > On 10/24/07, Benny Prijono <bennylp at pjsip.org> wrote:
> > > > >> Danny Brown wrote:
> > > > >>> I am using the auto answer function of the pjsua on win32. It
> answers
> > > > >>> like it should but I always get 1 audible ring. I would like to
> not
> > > > >>> get an audible ring at all. Is there a way to do this with the
> > > > >>> configuration file or what is the source file to change this
> behavior.
> > > > >> With "--auto-answer 200", pjsua will send 100 and followed
> > > > >> immediately by 200. Caller should not play any ring as no 180 is
> sent.
> > > > >>
> > > > >>   -benny
> > > > >>
> > > > >>
> > > > >> _______________________________________________
> > > > >> Visit our blog: http://blog.pjsip.org
> > > > >>
> > > > >> pjsip mailing list
> > > > >> pjsip at lists.pjsip.org
> > > > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> > > > >>
> > > > >
> > > > > _______________________________________________
> > > > > Visit our blog: http://blog.pjsip.org
> > > > >
> > > > > pjsip mailing list
> > > > > pjsip at lists.pjsip.org
> > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> > > >
> > > >
> > > > --
> > > > Benny Prijono
> > > > http://www.pjsip.org
> > > >
> > > >
> > > > _______________________________________________
> > > > Visit our blog: http://blog.pjsip.org
> > > >
> > > > pjsip mailing list
> > > > pjsip at lists.pjsip.org
> > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> > > >
> > >
> > >
> > > _______________________________________________
> > > Visit our blog: http://blog.pjsip.org
> > >
> > > pjsip mailing list
> > > pjsip at lists.pjsip.org
> > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> > >
> >
> > _______________________________________________
> > Visit our blog: http://blog.pjsip.org
> >
> > pjsip mailing list
> > pjsip at lists.pjsip.org
> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>



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