similar to: Zap channel(s), meetme and codecs/licences

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Hi, is there a way to use g729 in meetme? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070929/74f6e5d9/attachment.htm

We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

I haven't seen this appear on the list, so I thought I would resend it... Sorry for the repost if it did appear before... ----- Forwarded message from Michael George <george> ----- Date: Wed, 3 May 2006 21:48:09 -0400 From: Michael George <george> Subject: meetme conference latency degrades... To: asterisk-users@lists.digium.com We have recently started making more frequent use

I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly appreciated!

Here's the scenario: 1. I call out through * using a X100P card to somebody. Then I transfer them to a MeetMe conference and that all works. 2. After the conference is over everybody hangs up but "show channels" shows that the Zap/1-1 channel is still in use by MeetMe and the analog line is not freed up for re-use. Ever. Any clues? Thanks!

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? to whom ? how do I install them on asterisk etc ? thanks in advance , jl

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Hello !! I want to use MeetMe by G729. Can it be used if you buy a license? Please help me!!

Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the "old" way? Thanks in advance, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/6dd07c4b/attachment.htm