Displaying 20 results from an estimated 10000 matches similar to: "Jingle support - can we test the feature ?"
2012 Sep 20
1
chan_motif, xmpp, jabber, jingle
Hi all,
For one of my inverstigations it looks like i'm back to "square one"
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but
some suggested me to have a look at asterisk11,so i did...
I
2008 Oct 26
1
jingle/gtalk still very troubling
Hi!
I just tried to call a friend using jingle, but I got refused. Errorcode was
502, he tried to call me, heard it ringing once and then it stopped.
I used:
originate jingle/gtalk_account/friend at jabber.linuxlovers.at [application]
I'm registered to googletalk, but this should mean no harm, or should it.
Once I was able to receive a text-message from him, but couldn't
2006 Feb 19
3
Wildfire messsaging server
http://www.jivesoftware.org/
Is anyone running a wildfire messaging server on the same pc as their
asterisk server?
Is anyone specifically running it on an asterisk@home installation?
TIA,
Dean
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060219/539a4ccd/attachment.htm
2007 Nov 08
3
Asterisk as a SIP to XMPP Jingle voice gateway
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/
2006 Sep 12
2
Jabber server?
Hi everyone,
I am looking for a Jabber server for CentOS.
Any ideas?
--
"I never look back, it distracts from the now", Edna Mode (The Incredibles)
2007 Aug 28
3
Speex is the default codec for Jabber's Jingle VoIP
Just a heads-up, I received confirmation that Speex is now the default
codec for the Jabber's Jingle VoIP protocol.
While not the default in Google's Jabber, Speex has been reported to
work on Google Talk as well as of last year.
This information is not news breaking, but many people aren't aware of
it yet, so spread the word.
-Ivo
2008 Mar 28
1
jingle with Asterisk + PSTN
Hi All
I am developing a client that uses libjingle to do xmpp stuff with
ejabberd. I can also make audio calls between those clients. What I am
trying to archive now is to send calls to pstn using jingle. I was
told in the jingle-dev community that asterisk can do that.
Is there any way to send jingle audio calls to asterisk and will it
understand them ? If yes..can I forward those calls to PSTN
2007 Aug 28
4
Speex is the default codec for Jabber's Jingle VoIP
Peter Saint-Andre a ?crit :
> Ivo Emanuel Gon?alves wrote:
>> Just a heads-up, I received confirmation that Speex is now the default
>> codec for the Jabber's Jingle VoIP protocol.
>
> Which we hope to finalize soon for broader adoption. :)
That's good to hear. Are you supporting wideband or just narrowband?
Jean-Marc
2009 Nov 30
0
Asterisk and XMPP Jingle : testers needed
Dear community members,
I'm happy to announce that we now have code that allows you to use
your XMPP (Jabber) client like a softphone to place SIP or PSTN (or
whatever channel Asterisk supports) calls.
The XMPP clients that support Jingle that I and others have tested are :
- Pidgin (Linux, Ubuntu 9.10), version 2.6.2 : OK
- Empathy (Linux, Ubuntu 9.10), version 2.28.1.1 : OK
- Psi (Windows
2011 Mar 27
0
Jabber/Jingle to Google users via local XMPP server
Hi all,
All the examples I've come across seem to suggest configuring
jabber.conf/jingle.conf/gtalk.conf for a real Google account.
What about the scenario where the Asterisk server should connect to an
account on a private Jabber server and using Jingle (voice calling over
Jabber)?
e.g. for the domain widgets.com:
- there is a copy of ejabberd running on the same box as Asterisk, and
2008 Apr 21
0
Asterisk Jingle<->SIP GW Question
Dear All
I am using gtalk features with my own XMPP server "OpenFire"
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
So far so good. So if I want to call sip:1000 I call the xmpp account that
is bound to that account in extensions.conf.
2009 Jan 16
0
gtalk and jingle again...
Hello everyone!
I just installed the latest asterisk from svn. Now I'm retrying my luck with
gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not
sure if it helps or hurts.
I tried this:
call myself:
channel originate gtalk/gtalk_account/juliencoder at googlemail.com application \
Jack i(system:playback_1)o(system:capture_1)
I got some notes about a lot
2007 Aug 28
1
Speex is the default codec for Jabber's Jingle VoIP
Ivo Emanuel Gon?alves wrote:
> Just a heads-up, I received confirmation that Speex is now the default
> codec for the Jabber's Jingle VoIP protocol.
Which we hope to finalize soon for broader adoption. :)
> While not the default in Google's Jabber, Speex has been reported to
> work on Google Talk as well as of last year.
BTW, my contacts on the Google Talk team report that
2014 Jul 10
0
Unable to create Jingle session
Dear All,
I have different Asterisk Servers most of them are version 1.8 - I have
recently upgrade to Asterisk version 11 on 2 servers.
I have Jabber ( chan_gtalk ) configured on 1.8 version and it is working
within all 1.8 version servers.
I have XMPP ( chan_motif ) configured on 11 version and it is working with
all 11 versions servers.
When I try to call from version 11 ( usiing xmpp -
2014 Aug 09
0
chan_motif - Unable to create Jingle Session
Dear All,
I have different Asterisk Servers most of them are version 1.8 - I have
recently upgrade to Asterisk version 11 on 2 servers.
I have Jabber ( chan_gtalk ) configured on Asterisk 1.8 version and
it is working perfect
within all 1.8 version servers.
I have XMPP ( chan_motif ) configured on Asterisk 11 version and it
is working with
all 11 versions servers.
When I try to call from
2008 Oct 27
1
gtalk/jingle full report
Hello everyone!
Philippe, you told me to make a bugreport. Well, here it comes, I'm still
not sure, if tis is a bug or a miss-configuration.
So I've put up a collection of configurations/output/debug files from a
simple asterisk session testing the gtalk call.
You can download it here:
http://juliencoder.de/ap.txt
Or I can mail it, just tell me where and I'll attach it to
2011 Feb 10
2
Gtalk/Jabber Issue
OK, im pulling my hair out, everything looks configured right, deleted, and
started over, etc, etc. but can't seem to get this to work
Gtalk.conf
[general]
context=google-in
allowguest=yes
bindaddr=192.168.xxx.xxx
extenip=96.254.xxx.xxx
[guest]
context=google-in
disallow=all
allow=ulaw
allow=g729
connection=jp_jabber
jabber.conf
[general]
debug=yes
2008 Oct 17
1
[OT] RE: CELT 0.5.0 is out
Use Jingle, anyway Jingle kicks SIP on almost every aspect. Especially
on the way the standards are made.
Diana
David Hogan wrote:
>> I understand, but CELT would be useless for SIP if one can't
>>
> read/guess
>
>> correctly decoder configuration from the RTP data.
>>
>> One possible way to cope with this would be to have several CELT
>>
2006 Jan 08
1
JiveMessenger HOWTO
Any one out there done the JiveMesenger jabber server?
www.jivesoftware.com/messenger/
I want to get this running to then do the next step of tie-ing it in
to the * server for presence & callerID screen pops.
Pursued their site a bit but never found a HOWTO or anything that
looked relevant.
Appreciated.
Peace. Love. Linux.
Jason SJOBECK
ICQ 5579183
2011 Apr 16
5
Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: <iq from="+
17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="