similar to: Question about Management Interface

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

I have one single endpoint [inhouse](endpoint-basic) type=endpoint and one section like this [indentify] endpoint=inhouse match=X.Z.Y.X But when I call in from IP X.Z.Y.X, it does not match my peer. It tries to send the call the "default" context, and the number is not found. I know that I should have loaded this module, res_pjsip_endpoint_identifier_ip.so and it is loaded. In fact,

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

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,

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

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, is cdr logging using odbc buggy? I'im only getting an error "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.". Connecting with isql to the datasource given in cdr_odbc.conf works just fine, and using the database for sippeers also works. The documentation[1] is confusing because it says freeTDS is required and that you must not use multiple

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

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 >>

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.

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

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

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

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

On 12/07/2018 02:25 PM, Joshua C. Colp wrote: > On Fri, Dec 7, 2018, at 9:19 AM, hw wrote: >> >> Hi, >> >> is cdr logging using odbc buggy? I'im only getting an error >> "cdr_odbc.c:174 odbc_log: Unable to retrieve database handle. CDR failed.". >> >> Connecting with isql to the datasource given in cdr_odbc.conf works just >> fine,

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:

Hello Joshua, What will be example of endpoint configuration that not require authentication from specific ip ? volga629 On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:55 AM, volga629 at networklab.ca wrote: >> If understand correctly type=identify is more for sip trunk >> configuration ? >> >>

It is in the To: Header. -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] F?r Joshua Colp Skickat: den 19 februari 2018 11:58 Till: asterisk-users at lists.digium.com ?mne: Re: [asterisk-users] # converts to %23 On Mon, Feb 19, 2018, at 4:24 AM, Marcus Kvarsell wrote: > Hello, > > I have a broblem in

On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > Same problem with me. > > I downloaded the file in 2 different places and had the same error... An issue was filed for tracking this[1] and it will be resolved later today. [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL