search for: chan_motif

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 version 11 ( usiing xmpp - chan_motif ) to version 1.8 server it is always failed with following error =====================================================================================...

...icemail. 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 the defined user becomes online. >From jitsi I can send a text-message, which I see as I enabled "debug" in motif.conf (This is actually progress, as in 1.8.15.1 I saw...

I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the "-4" command line option to use IPv4 only just in

Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150117/f148cad7/attachment.html>

...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 same. Is the XMPP support only tuned for Google variation of XMPP/ICE/TURN, or is it supported for all open Jabber servers? I currently run 1.8 (before chan_motif) against ejabberd

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 - chan_motif ) to version 1.8 server it is always failed with following error ==========================================================================================...

Is it me or Google just blocked Asterisk's chan_motif? I get "violation of terms of service" audio message whenever I send a call. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130520/7f0148f3/attachment.htm>

I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s at xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de usuario XMPP ") in new stack motif.conf [jingle] context=xm...

...s on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

...[Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'app_speech_utils.so': /usr/lib64/asterisk/modules/app_speech_utils.so: undefined symbol: ast_speech_dtmf [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_motif.so': /usr/lib64/asterisk/modules/chan_motif.so: undefined symbol: ast_xmpp_client_send_message [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'func_aes.so': /usr/lib64/asterisk/modules/func_aes.so: undefined symbol: ast_aes_set_encrypt_key [S...

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

...ct 23 00:29 /usr/lib64/asterisk/modules/chan_iax2.so -rwxr-xr-x. 1 root root 41888 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_local.so -rwxr-xr-x. 1 root root 118144 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_mgcp.so -rwxr-xr-x. 1 root root 67424 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_motif.so -rwxr-xr-x. 1 root root 11936 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_multicast_rtp.so -rwxr-xr-x. 1 root root 44392 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_phone.so -rwxr-xr-x. 1 root root 755296 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_sip.so Any help appreciated. sea...

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? F...

Hi, I am on asterisk 12.6.0. Previously I was using 10.0.1 and for Gtalk I was using chan_gtalk and jabber configuration. But on 12.6.0 I tried to use chan_motif, asterisk starts consuming 100% cpu. From pstack trace I got it is because of ICE and to run motif ICE is necessary. Has anyone else seen this issue or any solution for this? I am using neither STUN nor TURN. I have just enabled ICE in rtp.conf. Regards Mayank Kumar Gour -------------- next part...

...ng [s at DialOut:15] Dial("DAHDI/1-1", "motif/8447/+1212xxxyyyy at voice.google.com,,rTt") in new stack [Mar 1 21:24:06] ERROR[2477][C-00000000]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? [Mar 1 21:24:06] ERROR[2477][C-00000000]: chan_motif.c:1820 jingle_request: Unable to create Jingle session on endpoint '8447' ???? any help appreciated. sean

Hi, Are there any thoughts about how "cpu-expensive" motif is? Does it only translate SIP <--> jingle (during call-setup) if so, impact will probably neglectible. or does asterisk remains constantly in between the data-stream? In that case, it might be something to pay serious attention to, when doing multiple call conversions simultaneously... hw

..._meetme: Fix channels lingering when hung up under certain conditions (Closes issue ASTERISK-20486. Reported by Michael Cargile) * --- Fix stuck DTMF when bridge is broken. (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) * --- Add missing support for "who hung up" to chan_motif. (Closes issue ASTERISK-20671. Reported by Matt Jordan) * --- Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. (Closes issue ASTERISK-20643. Reported by coopvr) * --- Fix chan_sip websocket payload handling (Closes issue ASTERISK-2074...

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? -- Chris -------------- next part

Hello; When I type make menuselect and finding the channels that has the sign XXX before it (this at the driver), how can I know the dependencies that are causing this conflict? Regards Bilal

...* ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits (Reported by Jeremy Lain??) * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir Cohen) * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with realtime peers (Reported by ibercom) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path (Reported by Corey Farrell) * ASTERISK-24378 - Release AMI connections on shutdown (Reported by Corey Farrell) * ASTERISK-24354 - AMI sendM...