[asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway

[Eric Chamberlain]

Philippe,

Thanks for the info.

--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Philippe Sultan
> Sent: Friday, November 09, 2007 2:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk as a SIP to XMPP Jingle voice
> gateway
> 
> Hi Eric,
> 
> > I'm looking for a SIP to XMPP Jingle voice gateway.
> >
> >
> >
> > I see that Asterisk has Jabber and Jingle support, but it looks like
> > Asterisk acts as a Jabber client.
> 
> Asterisk can connect as a client or component to a XMPP server. XMPP
> components are typically used as gateways between XMPP and other IM
> services such as MSN or Yahoo.
> 
> You can connect Asterisk to GoogleTalk's XMPP network as a client
> only, which will therefore be accessible through a presence
> subscription mechanism just like a usual client.
> 
> On the other hand, you can connect Asterisk as a component to your
> locally administered XMPP server, for example. A 'service discovery'
> request to the server will show the Asterisk server as being
> available.
> 
> > Are there any Jabber server solutions, where Jabber users can call SIP
> users
> > by using the SIP URI and vice versa?
> 
> Asterisk can be used to call Gtalk users from SIP phones, and vice
> versa. Configuration examples are given here :
> http://www.voip-info.org/wiki/view/Asterisk+Google+Talk
> The call configuration is handled in the Dialplan in that case.
> 
> If you need to place a call from a XMPP client to a SIP URI, you'll
> also have to find a client that's able to to so. I know that
> GoogleTalk and Jabbin both speak XMPP + Gtalk. However, the GoogleTalk
> client's user interface does not allow you to place a call to anything
> but another XMPP client from your buddy list, without offering the
> ability to enter either a SIP URI or phone number. A possible
> workaround was available here :
> http://bugs.digium.com/view.php?id=8659
> 
> As for Jingle, Asterisk tries to follow the latest set of
> specifications (code only available from SVN trunk), which are not
> completed yet.
> 
> Philippe
> 
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