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 <[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 > > _______________________________________________ > > 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
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