Have you figured out if asterisk is crashing or not? > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Scott Moseman > Sent: Friday, October 12, 2007 2:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] My G729 problem re-visited > > Gateway sends Asterisk an INVITE (using g729) Asterisk sends > Phone an INVITE (using g711 or g729) Phone sends Asterisk an > OK (using g711) Asterisk sends Gateway an OK (with no RTP > choice) Gateways ends the conversation > > I can setup the Phone to use g729 and it will reply with an > OK for g729, but the OK to the Gateway will still stay empty. > Only when I enable g711 on the Gateway will this work. I > have experienced this on > 2 different models of gateways so far. > > I included my config for both the Gateway and the Phone in my > original message, hoping that maybe I was configuring the > Gateway wrong in Asterisk? But no one has said anything so > I'm assuming its okay. > > Phone (g729) to Phone (g729) works > Phone (anything) to Gateway (g711) works Phone (anything) to > Gateway (g729) does NOT work > > I'm licensed for g729 (although I'm told I should not need it > for pass through). And it will transcode when the phone is > g729 and the gateway is g711. But for whatever reason I > can't use g729 on the gateway side of the calling process? > > Thanks, > Scott > > > > On 10/12/07, Power, Paul C. <[EMAIL PROTECTED]> wrote: > > > > 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: 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 >
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