That was actually a VM. Here's the real server (13ms). CLI> show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw - 3 - 1 2 2 1 3 13 - 15 2 - alaw - 3 1 - 2 2 1 3 13 - 15 2 - g729 - 5 4 4 4 4 3 5 - - 17 4 -
# dmesg | grep 'Xeon(TM)' CPU0: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 CPU1: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 Thanks, Scott On 10/12/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote: > > 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 > > _______________________________________________ --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