search for: jingle

Hi there, I work at an internet radio station. My management thought it might be a good idea to enhance our radio with targeted ads. Do you know of any streaming server that is capable of inserting targeted ads into the stream? So far I have only found Ando Media's Targeted Ad Injector (http://andomedia.com/) which claims to be open source but I can't find the source anywhere. Regards,

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 rece...

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...

...u are welcome to try it out and send feedback http://www.icecast.pwp.blueyonder.co.uk/icecast-2.3.2-kh-ads.tar.gz with this, each listener is allocated a group id (default 0) which the server will use to lookup a file to play. The number of groups is determined by the <mount> setting <jingle-groups>. At the moment the group is only set in auth url by the Jingle-group; header, although I do have a hardcoded switch to allocate the next id to the next new listener for testing. You essentially set the group to file mapping via admin eg /admin/jingle?mount=/stream&id=0&file...

...ne 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 downloaded and built 11-beta1. Edited (according to the asterisk11 wiki-page) extensions.conf, chan_motif.conf, jingle.conf and restarted. Same behavior, except for minor details. As soon as I start, ejabberd tells me that t...

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 XP), version 0.13 : Call establishes, but no sound (seems to be a problem with Speex) For the moment, one can only place calls from the XMPP cl...

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...

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

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

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

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 of traffic going...

Hi Chris, Thanks for the quick input. The harder the LLVM is, the harder it is for me to teach the course:-) Too many optimisations have been added, meaning I have to design many new projects. --- Jingling On Mon, Feb 23, 2004 at 09:41:13PM -0600, Chris Lattner wrote: > On Tue, 24 Feb 2004, Jingling Xue wrote: > > > I understand that LLVM is now available in the public domain. >

Jingling, I faced the same issue in using LLVM for an introductory compiler course this semester. The way I am (optimistically) addressing it is that I have given the students a tarball of LLVM containing most of LLVM but very few optimizations. In particular, we've only given them a few essential transformations that the front-end or lli need, and any transformations used by those

Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob.

...he xmpp account that is bound to that account in extensions.conf. However what do I have to do to make this work with PSTN numbers. I can just setup an entry + extensions for each pstn number I want to call. I know that I can parse the incoming number and send it to the PSTN with sip, however with jingle the number must be online already since jingle is presence based. So I must have a registered client for each number I want to call in the following format XMPP <-----------> SIP 1000 To Call 1000 //sip extension 1001 To Call 15461315461 //pstn num 1002 To Call 46456543213...

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, a...

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 Speex *is* the default for their voicemail f...

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. >...

...g error ========================================================================================== -- Executing [872 at longdistance:1] Dial("SIP/804-0000003d", "Motif/jitsi/ myfakename at jit.si") in new stack [Jul 5 16:52:21] ERROR[28270][C-0000002b]: chan_motif.c:1814 jingle_request: Unable to create Jingle session on endpoint 'jitsi' as no capable resource for target 'myfakename at jit.si' was found [Jul 5 16:52:21] WARNING[28270][C-0000002b]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'Motif' (cause 42 - Switching equip...

...ongdistance <http://lists.digium.com/mailman/listinfo/asterisk-users>:1] Dial("SIP/804-0000003d", "Motif/jitsi/myfakename at jit.si <http://lists.digium.com/mailman/listinfo/asterisk-users>") in new stack [Jul 5 16:52:21] ERROR[28270][C-0000002b]: chan_motif.c:1814 jingle_request: Unable to create Jingle session on endpoint 'jitsi' as no capable resource for target 'myfakename at jit.si <http://lists.digium.com/mailman/listinfo/asterisk-users>' was found [Jul 5 16:52:21] WARNING[28270][C-0000002b]: app_dial.c:2437 dial_exec_full: Unable to cr...