search for: russellbryant

List, Anybody have success with Digium's G729 codec and asterisk 1.6.0? Reading http://www.russellbryant.net/blog/index.php/2008/03/05/codec_g729-v34-builds-now-available/ is seems they are build for 1.6 and trunk. But all I could find / use is 1.4 builds from http://downloads.digium.com/pub/telephony/codec_g729/ Thoughts? PB

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme in dialplan: [some-context] exten => 123,1,Dial(Local/124 at some-context/nj) exten => 124,1,MeetMe(some-room,dM) So, the problem with voice quality was completely solved, BUT some customers have i...

...ello, I've read in Mantis that asterisk.conf's "internal timing" option could positively impact Asterisk behaviour during faxing ( http://issues.asterisk.org/view.php?id=16374). Before using it, I would be very pleased to read a line or two about its use. I've read http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/but I still a couple of questions. When you have a 1.6.1 server with a PSTN trunk, is this option of any use, as in my opinion, timing is then provided by PSTN ? Regards -------------- next part -------------- An HTML attachment was scrubb...

In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ /O

Hi, I was wondering if there's any sense in increasing audiobuffer above the minimal '2' in meetme, if every channel is already dejittered before (Local/.../nj - as described at: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) Will it help in anything, or just increase delay? Thanks, Stan

Hi, Russell's blog[1] is down and there are not much information about this any where else. Any one with more information about res_ais and how it is used? raj [1] http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/

On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote: > > Hello group, > > > > what is the preferred method to connect to asterisk cli over > > network? I need to run asterisk cli commands remotely. > > As others have mentioned: the manager interface is