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Hey All, For those of you that do not know me, my name is Matthew Fredrickson and I’m the project lead for the Asterisk project. First off, I wanted to thank all of you that contribute in various ways to the project – whether it be at a developmental level, answering questions on forums and mailing lists, contributing documentation, or just generally advocating for it within...

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan...

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

Anyone is using 1.6 in production?? Is it ready? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081107/df5bb63a/attachment.htm

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

...lists may > be in a situation similar as mine. > > > Thanks, > -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 > > Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit : > > Hey All, > > > > For those of you that do not know me, my name is Matthew Fredrickson > > and I’m the project lead for the Asterisk project. First off, I wanted > > to thank all of you that contribute in various ways to the project – > > whether it...

Just a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

...GRELOAD|NONE|NONE|NONE > > column types needs modification to something more appropriate > > can someone with confluence access ad info to > > https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ? Which info are you referring to? The table schema? Matthew Fredrickson > > > is there somebody using it in production? > thanks > > Dne 20/10/2016 v 10:16 marek cervenka napsal(a): > >> hi, >> >> is it possible log cel/queue_log to sqlite? >> >> via odbc? >> >> any experience? >> >> marek &gt...

...:39 +0000 [r4122] sruffell <sruffell at localhost>: * kernel/wct4xxp/base.c: Work around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for. 2008-03-14 16:39 +0000 [r3983-3990] Matthew Fredrickson <creslin at digium.com> * firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 * kernel/wct4xxp/base.c: When doing the ISR rewrite, forgot to include the vp...

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache...

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Hi all, I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. I've tried to unload various modules; nothing resolved the issue. Any suggestions? -- Mike Diehl

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

...sterisk 13.10 and save yourself a whole lotta headache. > > On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens at telenet.be > <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > Using Asterisk 12.8.2. > > > On 10-08-16 22:03, Matt Fredrickson wrote: > > My suggestion is to verify and debug against Asterisk 13 > first, and > then you can try backing down versions, rather than reverse. > WebRTC > is a rapidly moving target, and has required ongoing changes > that may &g...

Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287

On 10-08-16 08:52, Ludovic Gasc wrote: > > For WebRTC, I recommend you to use Asterisk 13+. > > Have a nice day. > > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > > > Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??

...e. We just updated the documentation for how to get CMP2K working on the wiki [1]. We'd love some feedback if you still have issues getting it setup so that we can improve the docs. [1] https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone Best wishes, Matthew Fredrickson

Hi, I just put together a little mixin to provide pseudo-RESTful services in camping apps. Basically, it looks for a hidden _verb field in form posts, and sets the @method to the supplied value (e.g. put or delete - which browsers don''t support). This lets you define put and delete methods in your controllers. Groovy. Code: http://pastie.caboo.se/22613 Is there any permanent repo of

In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg at mail.gmail.com>, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield <tony at softins.co.uk> wrote: > > I have some more investigation to do on this, but I wanted to see if anyone > > here had any insight into the issue I've run into. > > > >...