similar to: Google Voice

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

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

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 running asterisk 16 on Fedora 31. If I start asterisk as user asterisk, all goes well. But if I start asterisk from systemd: asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 sorcery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: config_options.c:710

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

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

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

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

Hi all,I hope to find a solution with the help of the list, I'm trying to get the status of my extensions with ejabberd , the idea is to visualize my users ejabberd incoming calls or missed. I'm testing with my operator extension with this code but only get the missed call notification does not show me where the call is coming. my piece of code [operadora] exten =>

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

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

Hi everybody, I've a matter with the queue annoucement with the "thereare", because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member => SIP/2098 and member => SIP/2099), the ivr don't gave me the range but It play the

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip*

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 -

Asterisk Project Security Advisory - AST-2014-011 Product Asterisk Summary Asterisk Susceptibility to POODLE Vulnerability Nature of Advisory Unauthorized Data Disclosure Susceptibility Remote Unauthenticated Sessions Severity Medium

Asterisk Project Security Advisory - AST-2014-011 Product Asterisk Summary Asterisk Susceptibility to POODLE Vulnerability Nature of Advisory Unauthorized Data Disclosure Susceptibility Remote Unauthenticated Sessions Severity Medium

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? For instance, for each sip-user an xmpp-user? When i skim