search for: res_xmpp

...the patch input. > > is the correct way to apply the patch or am I doing wrong? > > regardss > I'm confused this is not a patch, it's just garbage ;), I'm making a connection xmpp with asterisk and not connected, at the cli shows me the message every second: RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client ' RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when try...

Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com

2015-03-18 10:52 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > Hi list , this is a bug? > > > ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection > available when trying to connect client > > regardss > Hi , I'm trying to apply this patch from the source asterisk asterisk-11.16.0 and when I apply it shows me this message asterisk-11.16.0]#patch -p0 < refs patch: **** Only...

I have a multi-homed machine (quite a few IP addresses 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

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

...st at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. PJSIP Video on WebRTC Ast 13 (Gosmac) > 2. Re: res_xmpp.c:3468 xmpp_client_reconnect: (ricky gutierrez) > 3. Re: Asterisk 13 : SILK codec ? (Steve Murphy) > 4. Re: Asterisk switching bridge to native_rtp even with > direct_media=no (Matthew Jordan) > 5. Re: Asterisk 13 : SILK codec ? (Matthew Jordan) > 6. Problems p...

...k[1411]: [Jan 21 19:36:47] WARNING[1411]: loader.c:2381 load_modules: Some non-required modules failed to load. asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 load_modules: res_stun_monitor declined to load. asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 load_modules: res_xmpp declined to load. asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 load_modules: Declined modules which depend on res_xmpp: chan_motif asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 load_modules: chan_iax2 declined to load. asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]:...

Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work with PJSIP? The net effect here is that I want to get away from res_xmpp, if possible. Matt Hoskins | NPG Corp | Systems Architect

...Asterisk is susceptible to both portions of the vulnerability in different places. 1) The res_jabber and res_xmpp module both use SSLv3 exclusively, and are hence susceptible to POODLE. 2) The core TLS handling, used by the chan_sip channel driver, Asterisk Manag...

...Asterisk is susceptible to both portions of the vulnerability in different places. 1) The res_jabber and res_xmpp module both use SSLv3 exclusively, and are hence susceptible to POODLE. 2) The core TLS handling, used by the chan_sip channel driver, Asterisk Manag...

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

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>

...ned in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything I've tried to accomplish in that first priority has worked, but it never continues to an additional priority. Debug output looks like: [Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:2988 xmpp_pak_message: XMPP client 'testaccount' received a message [Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:3029 xmpp_pak_message: Deleted 1 messages for client testaccount from JID jabberclient at my.jabber.server [Aug 31 14:41:15] DEBUG[6954][C-00000000]: pbx.c:4410 pbx_extension_help...

...Asterisk. A device is automatically created and stored in the cache for each anonymous call that occurs; this is possible in the SIP and IAX2 channel drivers and through channel drivers that utilize the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). Attackers exploiting this vulnerability can attack an Asterisk system configured to allow anonymous calls by varying the source of the anonymous call, continually add...

...RID(num)}") same=> n,Dial(SIP/5001) same=> n,Hangup() same=> n(nodisponible), JabberSend(ejabberd,operadora at 172.16.8.59,"Llamada perdida de ${CALLERID(num)} ") same=> n,Hangup() look the log Oct 1 08:04:10] NOTICE[4789][C-00000028]: res_xmpp.c:1631 acf_jabberstatus_read: Resource alcides of buddy operadora at 172.16.8.59 was not found. -- Executing [0 at locales:1] Set("SIP/5002-00000029", "STATUS=7") in new stack -- Executing [0 at locales:2] GotoIf("SIP/5002-00000029", "0?disponible:nodispon...

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

...down a rabbit hole, does the mwi publish/subscription work for > non SIP phones? > > For instance, I have a single voicemail server, connected to multiple > asterisk boxes via SIP. On each of those servers, there are a mix of SIP > and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi > from the voicemail server to the endpoint servers. Would this type of > setup work with PJSIP? The net effect here is that I want to get away > from res_xmpp, if possible. > > Matt Hoskins | NPG Corp | Systems Architect > > > > ----------------------...

...d get this output (and the call fails with a "Declined" message on the calling phone). -- Executing [20005 at metalink:2] JabberSend("SIP/649EF376CA25-0000000c", "testaccount,user at my.jabber.server,Test") in new stack [Aug 21 15:42:54] WARNING[20469][C-0000000c]: res_xmpp.c:1752 xmpp_send_exec: JabberSend requires arguments (account,jid,message) I've tried putting in the full username and the [testaccount] for the first argument to JabberSend. I've tried just the username as well as the full user at my.jabber.server<mailto:user at my.jabber.server> i...

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

...at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolves the following security vulnerability: * AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability Asterisk is susceptible to the POODLE vulnerability in two ways: 1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their encrypted connections. 2) The core TLS handling in Asterisk, which is used by the chan_sip channel driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by default allow a TLS connection to fallback to SSLv3. This allows for...