similar to: Asterisk 13.3.0 IAX trunk issue with Yeastar

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 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 ===

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]

In particular, I'm comparing these two models: Yeastar NeoGate TG100 vs. Dinstar DWG2000-1G http://www.yeastar.com/products/NeoGate-TG100.asp http://www.dinstar.com/Product/Product_25.aspx?typeid=6 Wich model do you recommend me, Yeastar or Dinstar? Thanks in advance. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk -->

I am in the process of buying a TDM800 card from Yeastar ( http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20Card&cTypeName=1 ) Any one has tested this cards? How reliable are them? I am specially interested in the FXO/FXS module. -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux,

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

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

Toufic Khreish (Gmail) wrote: > 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

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 >

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

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

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>

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 investigate the problem of Asterisk 13 with video support on H264 ? Thank you. -------------- next part -------------- An

We have a small number of phones, when a call comes in we want all the phones that aren't in use to ring. Is there a simple way to test and see what phones are in use then ring the other phones? I tried some code like this: [zap] exten => s,1,Answer exten => s,2,ChanIsAvail(${DERRICK}) exten => s,3,SetVar,"EVERYONE=${DERRICK}" exten => s,4,ChanIsAvail(${DON}) exten

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 All, I have Cisco 7960's, Cisco 2950 Switch. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone is disconnects at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone seen this? Thanks for the help,

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Dale, Sorry for taking so long to answer, I've been traveling. Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dnoll at wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk