similar to: Asterisk 13.2.0 Video issues

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

Hello Matthew, I have compiled Asterisk 13.2 with the following compiler Flags enabled: DON'T_OPTIMIZE DEBUG THREADS BETTER_BACKTRACES My asterisk is running with the asterisk_script: root 24048 39.4 2.4 128564 50640 pts/1 Sl 00:02 2:21 /usr/sbin/asterisk -f -vvvg -c core show locks ======================================================================= === 13.2.0 ===

On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 > set debug trunk on >

I see that my asterisk is started with the -g option, the core file I cannot find on my system (find / -name core*) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, March 17, 2015 1:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Hi, Has anybody tried to connect Samsung Galaxy to Asterisk PBX to be able to make calls over VoLTE? Thanks a lot in advance! Best regards, Sevana http://www.sevana.biz -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150403/ac9b4a31/attachment.html>

If you take a look at the safe_asterisk shell script, usually located at /usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where the core files are located. If it's not located there, then you'll need to look at the Asterisk init script for the scripts location. I hope this helps. Regards; John -----Original Message----- From: asterisk-users-bounces at

On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems > with the format H264, Asterisk 12.8.1 compiled on the same hardware is > behaving very well for the same format H264 > > Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality. > > Could someone

Hello, Webrtc stopped after upgrading firefox from version 36 to version37. I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and firefox version 36 without any issues until firefox was upgraded to version 37. Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice. Could someone try

Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I?m sure I?m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me

Hello, I'm remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine). This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management. The former ADSL link has recently been replaced by a new 4G link (UMTS). I'm connecting to this box from a Debian Jessie/Gnome Terminal combo. With

On Mon, Mar 16, 2015 at 6:12 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > Hello Matthew, > > I have compiled Asterisk 13.2 with the following compiler Flags enabled: > DON'T_OPTIMIZE > DEBUG THREADS > BETTER_BACKTRACES > > > My asterisk is running with the asterisk_script: > root 24048 39.4 2.4 128564 50640 pts/1 Sl 00:02

On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > I see that my asterisk is started with the -g option, the core file I cannot > find on my system (find / -name core*) > I would suspect one of the following: (1) Asterisk is not actually crashing. (2) Something is deleting the core files. (3) The core files are hiding really, really

I can create an audio call file and specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course), and I do specify also the Codecs: h264,h263 etc... Asterisk reports: *File /tmp/video does not

Hi, I have tried Groundwire on IOS , and Android Alcatel (voice and video calls with asterisk 13.3) Also tried Bria on both OS in video and voice. Regards Toufic From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sevana Oy Sent: Friday, April 03, 2015 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hi all, I''ve noticed that facter version superior from epel do not display lsbmajdistrelease fact: # facter lsbmajdistrelease 5 # rpm -qa|grep facter facter-1.5.5-1.el5 # cat /etc/redhat-release Red Hat Enterprise Linux Server release 5.5 (Tikanga) # facter lsbmajdistrelease # cat /etc/redhat-release Red Hat Enterprise Linux Server release 5.5 (Tikanga) # rpm -qa|grep facter

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Hi, Is there any softphone or TAPI plug-in that allows one to dial from a web page? As you may know, Skype has a mechanism that converts phone numbers on a web page to a click-to-dial application. I'd like to use this but on a normal softphone (Bria, Xlite, other). Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

Hi all, my name is Arnau Bria and I''m a sys admin in a center where we must deal with hundred hosts. We''re currently working with quattor, but it''s too complex for our purposes, so I''m looking for new admin tool. I''ve been playing with CFengine for few days (2 or 3) and I''ve seen some limitations that makes me thing that CFE is not our