search for: jingles

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 receive a text-message from him, but couldn't

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

Micha? Jaszczyk wrote: > 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

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

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

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/

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

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.

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.

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

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

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 >>

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 -

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