Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours...
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Scott Moseman > Sent: Friday, October 12, 2007 10:22 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] My G729 problem re-visited > > No ideas on this one from anyone? I suppose I'm going to > need to pay for some Digium support because this is a really > unusual problem. > Does anyone else have a gateway that speaks g729 to Asterisk > and works? For whatever reason, Asterisk refuses to reply > back to any of my gateways using g729. Phone (g729) to phone > (g729) works. Phone > (g729) to Asterisk to gateway (g711) works. But attempt g729 > between Asterisk and a gateway and it fails -- every time. > Asterisk responds to the gateway but never includes any > codecs in the packet, unless it's g711. My configurations > are shown below. > > Thanks, > Scott > > > On 9/26/07, Scott Moseman <[EMAIL PROTECTED]> wrote: > > > > Ok, I built a test system to duplicate my problem and > provide myself a > > platform that I can mess around with to try and break any features. > > My problem is G729 pass-through from a gateway to a phone. > I think I > > even have transcoding working, which makes me more confused > on what's > > wrong with my pass-through. It must be a configuration issue. > > > > The basics... > > > > *CLI> core show version > > Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux > > > > *CLI> show modules like 723 > > Module Description Use Count > > codec_g723.so G.723.1 Coder/Decoder 0 > > format_g723.so G.723.1 Simple Timestamp File Format 0 > > > > *CLI> show modules like 729 > > Module Description Use Count > > codec_g729.so G.729 Coder/Decoder 0 > > format_g729.so Raw G729 data 0 > > > > *CLI> show translation > > [truncated] > > g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex > ilbc g726 g722 > > ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - > > g729 5 2 2 2 2 2 1 3 - - 11 2 - > > > > The configuration... > > > > [gateway] > > type=friend > > host=gateway > > context=default-inbound > > disallow=all > > allow=g729 > > > > [phone] > > type=friend > > context=sip > > host=dynamic > > username=phone > > secret=scott > > dtmfmode=RFC2833 > > disallow=all > > allow=g729 > > callerid=Scott > > qualify=yes > > canreinvite=no > > > > exten => 1266,1,Dial(SIP/[number],30,t) exten => 1266,2,Congestion > > > > exten => 1266,1,Dial(SIP/[number],30) > > exten => 1266,2,Congestion > > > > (The same results using both of the above dialplans...) > > > > The environment... > > > > PSTN -> Gateway -> Asterisk -> Phone > > > > What I'm seeing works... > > > > With the gateway setup to send both G711 and G729, it sends > an INVITE > > which includes both G711 and G729 codecs. Asterisk sends an > INVITE to > > my phone with only G729. The call is made and there's a > conversation > > in G711 with the gateway and G729 with the phone. I assume > this means > > Asterisk is transcoding. > > > > What I"m seeing fails... > > > > With the gateway setup to send only G729, it sends an INVITE to > > Asterisk which includes only G729. Asterisk send an INVITE to the > > phone using G729, too. The 200 OK from the phone to the Asterisk > > includes G729. The 200 OK going from Asterisk to the > gateway doesn't > > include ANY codec. The call is dropped the moment I pickup > the phone > > to answer the call. > > > > My question... > > > > Why does Asterisk not want to respond to my gateway in G729? > > Even if the gateway requests it, Asterisk seems to just ignore it. > > From the transcoding call, and phone to phone G729 calls, I > have proof > > that Asterisk knows how to handle G729 calls. > > > > Where do I go from here??? > > > > Thanks, > > Scott > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users