similar to: jingle with Asterisk + PSTN

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

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

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

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

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.

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

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 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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Guys Ive got an internet cafe on which I have a debian sarge box running. The Debian box acts as a gateway and it has masquerading on. I have 40 client PC and i do not want to assign more than 64k per pc for upload and the same is true for download too. Ive done alot of research and Ive read tutorials about CBQ and HTB. I found that CBQ.init is

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.

Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the

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

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

oops, i forgot to reply to the list :-/ Début du message réexpédié : > De : Vincent Dautremont <vdautrem@ulb.ac.be> > Date : 30 août 2007 16:58:26 GMT+02:00 > À : Ali Jawad <alijawad1@gmail.com> > Objet : Rép : [LARTC] Priotirize SSH Traffic > > try that > #tc qdisc add dev eth0 root handle1: prio > # tc filter add dev eth0 protocol ip parent 1: prio 1 u32

Hi I do have asterisk running in real time I do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [999alijawad at a2billing:1]