So, that is being made for * ? or SIP too? :)
Well, google has plans for SIP support. I guess it should be any time soon as they promised it last August I beleive.
Also, people on their developers board already coded the bridge for SIP (housemade stuff); I just don't have the time to go over it. That's why I was wondering.
Thanks for the feedback.
From: Shidan <[EMAIL PROTECTED]>
To: "B Nik" <[EMAIL PROTECTED]>
CC: [email protected]
Subject: Re: [on-asterisk] XMPP and SIP bridge
Date: Sat, 25 Feb 2006 15:58:47 -0500
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I take that back, actually I was just informed that Mark and the folks
at Digium have personally begun working on xml and jabber support in
asterisk, so a chan_jingle should be a reality soon.
----
Shidan
On 2/25/06, Shidan <[EMAIL PROTECTED]> wrote:
> You will have to wait untill google releases their SipProxy class as
> part of libjingle. Even then you will have to write/wait till someone
> writes a proxy using some other framework as no one will use expat in
> * in the near future.
>
> -------
> Shidan
>
> On 2/25/06, B Nik <[EMAIL PROTECTED]> wrote:
> >
> >
> > Hello,
> >
> > Google's Talk uses XMPP protocol for authentication and PCMU, iLBC, G723
> > codecs. Does anyone know of any apps out there which would bridge this with
> > my SIP server?
> >
> > Thanks,
> >
> > Bruce
> >
> >
> > ________________________________
> > From: "Tony Cowling" <[EMAIL PROTECTED]>
> > To: <[email protected]>
> > Subject: [on-asterisk] vonage and or generic x100p
> > Date: Thu, 26 Jan 2006 09:16:16 -0500
> > MIME-Version: 1.0
> > Received: from iran.unitycode.org ([205.207.27.18]) by
> > bay0-mc11-f12.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Thu, 26
> > Jan 2006 06:16:55 -0800
> > Received: (qmail 14928 invoked by alias); 26 Jan 2006 14:16:20 -0000
> > Received: (qmail 8880 invoked by uid 0); 26 Jan 2006 14:16:19 -0000
> >
> >
> >
> >
> > Hoping you can help.
> >
> >
> >
> > I have a couple of x100p generic cards in place, one of which is hooked to
> > my vonage line.
> >
> >
> >
> > I notice if you call the vonage number and get passed to voice mail on the
> > asterisk server there is a lot of back round noise.
> >
> >
> >
> > It would appear that the noise is only there when the is no input sound from
> > the receiving end if this makes sence.
> >
> >
> >
> > For example if you have a normal conversation on the vonage line, the back
> > round noise seems to die down but if I where to stay silent the noise would
> > build up to a rather annoying level. It would seem that the vonage line is
> > somewhat to blame for this however it would seem that passing through the *
> > box seems to amplify it.
> >
> >
> >
> > Does the quality of the x100p have something to do with this?
> >
> >
> >
> > Is there anything that can be done to help cut down the noise somehow by
> > generating a small amount of input from * for example when no-one is
> > physically receiving the call?
> >
> >
> >
> > Looking forward to your replies. Tony.
> >
> > ________________________________
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>
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