Philippe, Thanks for the info.
-- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] 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 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users