<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.3790.4064" name=GENERATOR></HEAD>
<BODY>
<DIV dir=ltr align=left><SPAN class=725032818-15102007><FONT face="Courier New"
color=#0000ff size=2>Try the Prescott version of the G729
.so.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=725032818-15102007><FONT face="Courier New"
color=#0000ff size=2>That one is made for xeon's.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Mike
Lynchfield<BR><B>Sent:</B> Friday, October 12, 2007 2:36 PM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] My G729 problem re-visited<BR></FONT><BR></DIV>
<DIV></DIV>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="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">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 <A
href="http://www.api-digital.com--">http://www.api-digital.com--</A><BR><BR>asterisk-users
mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">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 <BR>1.877.807.VOIP (8647)<BR>1.514.312.7030 </BLOCKQUOTE></BODY></HTML>