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Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

...d by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and jd at...

Hello all, I am new in asterisk configuration. I want to configure a Radius server to authenticate the sip users of asterisk. I have trying to use the next document: http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_ authentication.html Can you help me? Regards, Sergio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c4f51199/attachment.htm

hi, how to add radius support to asterisk 1.4.5? i do make menuselect and i do not see any module or option related to radius, pam, authenticacion or similar. any ideas? thanks

Friends, I will be attending FOSDEM in Brussells Feb 23-24. Anyone else? Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be there, so we could have a SIP/XMPP/Asterisk ad hoc meeting :-) On Thursday, Feb 21, I will be in Utrecht, Netherlands for the free Open Telephony conference at Media Plaza. There's still seats available and a really good talk about ENUM with Pat...

I am upgrading to raills 2.3.2. I’m having problems with AjaxScaffold which has been deprecated. I am a linux novice. The web site http://activescaffold.com/ has the following command: script/plugin install git://github.com/activescaffold/active_scaffold.git I tried the above and also ruby script/plugin … Nothing happens. Any ideas?

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 asterisk-xmpp account...

...for speakers with interesting ideas or stories - new applications with combinations of VoIP, IM, location, multimedia or just clever solutions with text messaging integrated with the new platforms... The conference will have a lot of interesting people as speakers. Both myself and Philippe Sultan (INRIA, France) will participate from the Asterisk developers group. If you have an idea, drop me an e-mail at oej at edvina.net At the end of the conference, on friday April 4th, I will run an Asterisk Community Meeting for everyone working with Asterisk, building systems, hacking, porting...

Hello, I would like to know, if in Asterisk version 1.6.0-beta9 is it possible to use Radius as an external authentication server. Thank you for your help Ciao Salvo

How do I use fastercsv to just read the header row and not the whole file? I need to read the header row first and not the whole csv file initially in the web application so I can map the input csv columns to the db columns. Once the mapping is complete, a rake task will than process the whole file. I don’t want to read the whole file to save processing time. How do I just read the header row

...word to the requesting LDAP client / asterisk server? This article made me wonder whether this might not that easily possible with LDAP due to the nature of the SIP protocol (i.e. it's challenge-response handshake) and that I'd need to use RADIUS instead: https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html Cheers, Linus

...gn before 0.7131 in preZ sim1<-dccsim(ADCC, n.sim =2345,n.start =100 , m.sim = 1, startMethod = "unconditional",preZ = - 0.7131) # extract covariance sim1_cov<-as.matrix(rcov(sim)[1,2,]) Please feel free to reply that this approach is correct to proceed. Thanks. Best regards, sultan [[alternative HTML version deleted]]

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

...I'm planning to migrate my site to a new server, so I prefer not to have new uploads just yet. Constructive feedback will be most appreciated! [1] http://lists.xiph.org/pipermail/theora/2009-August/002678.html [2] http://www.theora.org/downloads/ [3] http://www.basilgohar.com/blog/2009/06/05/sultan-mizan-zainal-abidin-stadium-aka-gong-badak-stadium-collapses [4] http://docs.fedoraproject.org/install-guide/f11/en-US/html/s1-rescuemode-booting-single.html -- Basil Mohamed Gohar abu_hurayrah at hidayahonline.org http://www.basilgohar.com/blog basilgohar on irc.freenode.net GPG Key Finger...

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

...hey are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at systems at quinnox.com Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, prin...

Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny

I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system,

...ble> <tr> <th>topics</th> </tr> <tr> <td> <span><a href="/forums/books/topics/most-popular-books- right-now" rel="bookmark">most popular books right now</a> by sultan 2 minutes ago - sticky</span> </td> </tr> </table> </div> <div id="footer"> <p>&copy; 2008 mysite. all rights reserved.</p> <ul> <li><a href="/admin">adm...

...pported xwayland: don't fall back to wl_drm with explicit modifier xwayland: use drmDevice to compare DRM devices Allow disabling the SHAPE extension at runtime xwayland: use gbm_bo_create_with_modifiers2() build: set _GNU_SOURCE when checking for SO_PEERCRED Sultan Alsawaf (11): modesetting: Remove redundant GLAMOR_HAS_GBM #ifdef from ms_do_pageflip modesetting: Pass reference CRTC pointer to ms_do_pageflip modesetting: Pass CRTC pointer to TearFree flip handlers modesetting: Fix memory leak on ms_do_pageflip error modesetti...