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.
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 <scmoseman at gmail.com> 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 >
How do you get 11ms translation time on ulaw 729 ? we have 12ms and its dual xeons 2.6.. On 9/26/07, Scott Moseman <scmoseman at gmail.com> 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 > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --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 >-- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071012/b3d6dd45/attachment.htm