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

Discussion:

Eric Chamberlain

2007-11-08 23:39:05 UTC

Hello,

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.

Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa?

--

Eric Chamberlain, CISSP

Chief Technical Officer

Voxilla - http://voxilla.com/

Geoff Jacobs

2007-11-09 00:04:58 UTC

Permalink

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body
style="font-family: Verdana; font-size: 13px; color: rgb(46, 52, 54);">
 
Openfire has a SIP soft phone plugin (and an AsteriskIM plugin -
different functionality though) 
<a href="http://www.igniterealtime.org/">http://www.igniterealtime.org/</a> 
 
 
Eric Chamberlain wrote:
<blockquote id="mid_d3ad7384960d144fbe9ccf6df4c0ebdd_voxilla_com"
cite="mid:***@voxilla.com" type="cite">
<style>

</style>
<div class="Section1">
Hello,<o:p></o:p>
<o:p> </o:p>
I’m looking for a SIP to
XMPP Jingle voice gateway. <o:p></o:p>
<o:p> </o:p>
I see that Asterisk has
Jabber and Jingle support, but it
looks like Asterisk acts as a Jabber client.  <o:p></o:p>
<o:p> </o:p>
Are there any Jabber
server solutions, where Jabber users
can call SIP users by using the SIP URI and vice versa?<o:p></o:p>
<o:p> </o:p>
--<o:p></o:p>
Eric Chamberlain, CISSP<o:p></o:p>
Chief Technical Officer<o:p></o:p>
Voxilla -
<a class="moz-txt-link-freetext" href="http://voxilla.com/">http://voxilla.com/</a><o:p></o:p>
<o:p> </o:p>
</div>
<pre wrap="">
<hr size="4" width="90%">
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre>
</blockquote>
</body>
</html>

Mik Cheez

2007-11-09 00:43:00 UTC

Permalink

Openser has "SIP-to-XMPP Gateway" and "JABBER IM and PRESENCE
interconnection" modules. I used the XMPP module with Google Talk and
Asterisk a while back. It resulted in a segmentation fault, but again
that was a long time ago.

Post by Eric Chamberlain
Hello,
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.
Are there any Jabber server solutions, where Jabber users can call SIP
users by using the SIP URI and vice versa?
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

Philippe Sultan

2007-11-09 10:39:13 UTC

Permalink

Hi Eric,

Post by Eric Chamberlain
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.

Post by Eric Chamberlain
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

Eric Chamberlain

2007-11-09 15:29:41 UTC

Permalink

Philippe,

Thanks for the info.

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

-----Original Message-----
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,

Post by Eric Chamberlain
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.

Post by Eric Chamberlain
Are there any Jabber server solutions, where Jabber users can call SIP

users

Post by Eric Chamberlain
by using the SIP URI and vice versa?

Asterisk can be used to call Gtalk users from SIP phones, and vice
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
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
http://lists.digium.com/mailman/listinfo/asterisk-users