similar to: chan_motif / res_xmpp problems

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

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

2017-09-07 15:04 GMT+02:00 Sami Ketola <sami.ketola at dovecot.fi>: > > > On 7 Sep 2017, at 16.03, Antoine Nguyen <ngu.antoine at gmail.com> wrote: > > > > 2017-09-07 14:29 GMT+02:00 Aki Tuomi <aki.tuomi at dovecot.fi>: > > > >> > >> > >> On 07.09.2017 15:26, Antoine Nguyen wrote: > >>> Hi all, > >>>

2017-09-07 14:29 GMT+02:00 Aki Tuomi <aki.tuomi at dovecot.fi>: > > > On 07.09.2017 15:26, Antoine Nguyen wrote: > > Hi all, > > > > I've just upgraded my server from debian 8 to debian 9 and I now > encounter > > a strange issue. I'm using prosody (XMPP server), configured to > > authenticate against dovecot using SASL and a unix socket.

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

Hi all, I've just upgraded my server from debian 8 to debian 9 and I now encounter a strange issue. I'm using prosody (XMPP server), configured to authenticate against dovecot using SASL and a unix socket. This setup was working fine on debian 8. Now, each time I try to connect using my XMPP client, the authentication fails and I see the following error in logs: auth: Error: BUG:

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload => res_jabber.so noload => chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side

2015-03-18 11:13 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > 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 garbage was found in the patch input. > > is the correct way to apply the patch or am I doing wrong? >

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

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

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

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

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

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 -

Hi, I'm currently installing and configuring CentOS 7 on a public server. The machine will host a few small-to-midsize projects that are currently running on a handful of Slackware servers: public library databases, our public school's agenda, a small webradio, OwnCloud for myself and a local non-profit, etc. Until recently I've mostly used self-signed SSL certificates for stuff

* Antoine Nguyen [2017-09-07 15:13]: > 2017-09-07 15:04 GMT+02:00 Sami Ketola <sami.ketola at dovecot.fi>: > > > > > > On 7 Sep 2017, at 16.03, Antoine Nguyen <ngu.antoine at gmail.com> wrote: > > > > > > 2017-09-07 14:29 GMT+02:00 Aki Tuomi <aki.tuomi at dovecot.fi>: > > > > > >> > > >> > > >>

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

> On 7 Sep 2017, at 16.03, Antoine Nguyen <ngu.antoine at gmail.com> wrote: > > 2017-09-07 14:29 GMT+02:00 Aki Tuomi <aki.tuomi at dovecot.fi>: > >> >> >> On 07.09.2017 15:26, Antoine Nguyen wrote: >>> Hi all, >>> >>> I've just upgraded my server from debian 8 to debian 9 and I now >> encounter >>> a

Le 03/05/2020 ? 08:44, Benson Muite a ?crit?: > They have rpms: > > https://download.jitsi.org/jitsi/rpm/ That's the Jitsi desktop application. > > and also a scalable installation: > > https://github.com/jitsi/jitsi-meet/blob/master/doc/scalable-installation.md That's Debian-specific. > > Which one do you need? None. I need a comprehensive

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything