similar to: Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!

Friends, a new dialplan application is now available for testing : http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ The corresponding feature request is located here : http://bugs.digium.com/view.php?id=12569 What can you do with it? Well, a direct usage of this application is to make an easy to use GoogleTalk voice gateway out of Asterisk. Here is an example (assuming the

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 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, Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) about Jabber, something the Asterisk community is becoming more interested in by the day. Join us to learn more about Jabber and SIP or to share your knowledge and experience. As always, the VUC discussion includes people from very diverse backgrounds, so

Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =

Hi All, I am doing some research on the intergration of Jabber and Asterisk. I have tried Jabber Client Mode. It's cool and works fine. But there's few information on the Component Mode. What's the difference between these two mode? I finished the configuration on jabber.conf and I am using openfire. What shall I do in the openfire to make them intergrated? Looks forward to your

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

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

I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount at my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console:

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client at theirdomain@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server /

I apologize for my poor English..... So, i don't really understand 'how to' realize thus.... When you use the cmd Dial and want to get $ from caller channel to callee (or callee channel from caller), which way is the right way ? Sorry, i've take a look to the wiki and asterisk code and is'nt limpid (for me) use Macro in the dial cmd.... hum it's on the caller side and i

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

Hello all, I am having trouble getting gtalk to work with my account which is not using a gmail.com email address. When I do this there an error from the Jabber module: [Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER: Node Error [Feb 3 20:51:17] WARNING[6286]: res_jabber.c:1495 aji_recv_loop: JABBER: Got hook event. JABBER: gtalk_account INCOMING:

Hey everybody. Another XMPP+Asterisk example: http://www.mundoopensource.com.br/en_page_send-xmpp-message-extensions-logged-asterisk-queue/ []s Marcelo H. Terres mhterres at gmail.com Openfire-BR mailing list's owner IM: marcelo at jabber.mundoopensource.com.br http://www.mundoopensource.com.br http://offtopicsandfun.blogspot.com http://biertasters.blogspot.com http://twitter.com/mhterres

Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is

Hi, Do you really think Presence should be used to forward call to voicemail ? My feeling is forwarding incoming calls to voicemail should remain a different task as you could wish to remain unavailable for chat and still reachable by phone. As I can't see a way to define Presence status such as "unavailable for chat and phones", "unavailable for chat but available for

Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli

I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this

Asterisk buddies! If you have open issues in the bug tracker, please help us with providing fast responses. All developers are working real hard to close bugs pending the new release, so we kindly ask you for fast responses on our questions in the bug tracker. The quicker the better and we'll get 1.2 out of the door sooner. If you have new ideas, feature requests, thoughts - please keep the