How do you get 11ms translation time on ulaw 729 ?<br><br>we have 12ms and its dual xeons 2.6..<br><br><div><span class="gmail_quote">On 9/26/07, <b class="gmail_sendername">Scott Moseman</b> <<a href="mailto:scmoseman@gmail.com">
scmoseman@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Ok, I built a test system to duplicate my problem and provide myself
<br>a platform that I can mess around with to try and break any features.<br>My problem is G729 pass-through from a gateway to a phone. I think<br>I even have transcoding working, which makes me more confused on<br>what's wrong with my pass-through. It must be a configuration issue.
<br><br>The basics...<br><br>*CLI> core show version<br>Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux<br><br>*CLI> show modules like 723<br>Module Description Use Count<br>codec_g723.so G.723.1 Coder/Decoder 0
<br>format_g723.so G.723.1 Simple Timestamp File Format 0<br><br>*CLI> show modules like 729<br>Module Description Use Count<br>codec_g729.so G.729 Coder/Decoder 0<br>format_g729.so Raw G729 data 0<br><br>*CLI> show translation
<br>[truncated]<br>g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722<br>ulaw 5 2 - 1 2 2 1 3 7 - 11 2 -<br>alaw 5 2 1 - 2 2 1 3 7 - 11 2 -<br>g729 5 2 2 2 2 2 1 3 - - 11 2 -<br><br>The configuration...
<br><br>[gateway]<br>type=friend<br>host=gateway<br>context=default-inbound<br>disallow=all<br>allow=g729<br><br>[phone]<br>type=friend<br>context=sip<br>host=dynamic<br>username=phone<br>secret=scott<br>dtmfmode=RFC2833<br>
disallow=all<br>allow=g729<br>callerid=Scott<br>qualify=yes<br>canreinvite=no<br><br>exten => 1266,1,Dial(SIP/[number],30,t)<br>exten => 1266,2,Congestion<br><br>exten => 1266,1,Dial(SIP/[number],30)<br>exten => 1266,2,Congestion
<br><br>(The same results using both of the above dialplans...)<br><br>The environment...<br><br>PSTN -> Gateway -> Asterisk -> Phone<br><br>What I'm seeing works...<br><br>With the gateway setup to send both G711 and G729, it sends
<br>an INVITE which includes both G711 and G729 codecs. Asterisk<br>sends an INVITE to my phone with only G729. The call is made<br>and there's a conversation in G711 with the gateway and G729<br>with the phone. I assume this means Asterisk is transcoding.
<br><br>What I"m seeing fails...<br><br>With the gateway setup to send only G729, it sends an INVITE<br>to Asterisk which includes only G729. Asterisk send an INVITE<br>to the phone using G729, too. The 200 OK from the phone to
<br>the Asterisk includes G729. The 200 OK going from Asterisk to<br>the gateway doesn't include ANY codec. The call is dropped the<br>moment I pickup the phone to answer the call.<br><br>My question...<br><br>Why does Asterisk not want to respond to my gateway in G729?
<br>Even if the gateway requests it, Asterisk seems to just ignore it.<br>From the transcoding call, and phone to phone G729 calls, I have<br>proof that Asterisk knows how to handle G729 calls.<br><br>Where do I go from here???
<br><br>Thanks,<br>Scott<br><br>_______________________________________________<br><br>Sign up now for AstriCon 2007! September 25-28th. <a href="http://www.astricon.net/">http://www.astricon.net/</a><br><br>--Bandwidth and Colocation Provided by
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Mike<br>Sales Manager<br><a href="http://www.voicemeup.com">http://www.voicemeup.com</a><br>Making it happen
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