Re: [asterisk-users] Specifying a different codec for meetme

Right. I fully understand that. The preference for g.711 in this case is that the transcoding is significantly cheaper (as reflected in the information obtained from core show translation). Thanks, --ag On Thu, Jul 17, 2008 at 6:45 AM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Artie Gold

Re: [asterisk-users] Specifying a different codec for meetme

Yes. Exactly. Thanks! --ag On Thu, Jul 17, 2008 at 9:01 AM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Artie Gold wrote: Right. I fully understand that. The preference for g.711 in this case is that the transcoding is significantly cheaper (as reflected in the information obtained from

Re: [asterisk-users] Specifying a different codec for meetme

Perfect! (Yes, we're using sip phones). Many, many thanks! --ag On Thu, Jul 17, 2008 at 9:38 AM, Mark Michelson [EMAIL PROTECTED] wrote: Artie Gold wrote: Yes. Exactly. Thanks! --ag If you're using SIP phones, you can set the SIP_CODEC channel variable in the dialplan prior

[asterisk-users] Specifying a different codec for meetme

to the general case where g.729 outperforms. Many thanks, --ag -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] Meetme replacement with native 729 support

Folks: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Many thanks, --ag -- Artie Gold F4W, Inc. ___ -- Bandwidth

Re: [asterisk-users] Meetme replacement with native 729 support

That makes sense -- thanks! --ag On Tue, Jul 15, 2008 at 1:59 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating

Re: [asterisk-users] Incoming

sip Thanks, --ag On Tue, Jul 15, 2008 at 8:39 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Artie Gold wrote: Folks: This is my first post, so please let me know if I transgress in any way... In updating to 1.4.21 recently, we've

[asterisk-users] Incoming

, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. Many thanks, --ag -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22

[asterisk-users] SIP timing out over satellite connection on 1.4.21 (works with 1.4.18.1)

insight would be appreciated. Many thanks, --ag [This is an edited repost with an improved subject line -- thanks to Steve Edwards] -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September

Re: [asterisk-users] Incoming

This is a quite promising idea. Many thanks. I'll post my results to the list... Cheers, --ag On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 11 July 2008 09:17:37 Artie Gold wrote: In updating to 1.4.21 recently, we've encountered a problem, when