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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

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

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 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, 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

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

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>

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603 at DLPN_AlDimnaDialPlan:601] Dial("SIP/601-00000002", "SIP/603") in new stack [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433 dial_exec_full: Unable to

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

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>: > Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >>

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, September 23, 2015 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04 On Wed, Sep 23, 2015 at 5:43 PM, Ryan, Travis <RyanT at

wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark

Hi all, Is there any native way to save endpoint statuses to database? I use asterisk 13 with pjsip and realtime, and didn't found proper way. I read that there is config parameter in sip.conf: rtupdate=yes. But how can I do that with pjsip? Or I should use sip.conf with pjsip simultaneously. Or is there any kind of hooks, which allows make custom action on endpoint status change. Thanks.

Ludovic Gasc wrote: > Hello, > > same sip username with realms is possible with Asterisk ? > I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and > now, Asterisk crashes. Did PJSIP crash in general (it's usually a build problem if that happens) or was it when you were experimenting with different realms and such? -- Joshua Colp Digium, Inc. | Senior

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I

Is the list going to be the same after sangoma take over digium? On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp <jcolp at digium.com> wrote: > On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > > that forum supposed to supersede this mailing list ? > > Both remain available but the community

> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 6:22 PM > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:17 PM, Ryan, Travis wrote:

On Sat, Mar 11, 2017, at 11:50 AM, Kseniya Blashchuk wrote: > Hey guys, any thoughts on that? Probably a bug or is it a default > behavior? I'd suggest providing the configuration to make sure it is correct. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

Sonny Rajagopalan wrote: > I receive a TCP ack back from that port (5060; owned by Asterisk) > --confirmed by wireshark on the Asterisk server. That's from Wireshark, but what is Asterisk seeing? If Asterisk doesn't show the connection or the traffic then something else is up (firewall, etc). Try to isolate things further, start from Asterisk itself. -- Joshua Colp Digium, Inc.