In article <caa_1sggvs-i_swslx77kgc3md1pre9br_s9vfxz-ryd_cqs...@mail.gmail.com>, s m <[email protected]> wrote: > > thanks Asghar, but are you sure? my two endpoints -which are soft-phones- > have g729 codec but my asterisk on middle system has not any module for > g729 codec. i think i should get module g729 for my middle system in order > to pass calls with g729 codec. isn't it true?
No, it's not. Asterisk can pass G.729 frames transparently from one endpoint to another without using a codec. However, that does limit the functionality of Asterisk a bit. Asterisk does need the G.729 codec any time it needs to understand the contents of a G.729 frame, in order to do any of these: 1. Play a sound (unless you have installed the g729-format sounds). 2. Save voicemail (unless you have configured voicemail ONLY to save g729 format). 3. Detect in-band DTMF (you should be using RFC2833 instead of inband anyway). 4. Meetme or Confbridge conferencing (needs to convert to linear in order to mix). 5. Play music on hold (unless you have installed only g729-format music). 6. Bridge a call to a device that cannot do G.729 format or a non-VoIP channel. There may be others I haven't thought of. So I guess if you have only G.729-capable devices, a G.729-capable SIP trunk, install the G.729 sounds and music and configure the system correctly, the only thing you can't do is conferencing. However, I don't see any real need to use G.729 unless you are severely limited on network bandwidth. It seems to be a solution to a historical problem that has largely gone away nowadays. You will get better quality with G.711 at least. Cheers Tony -- Tony Mountifield Work: [email protected] - http://www.softins.co.uk Play: [email protected] - http://tony.mountifield.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
