On 11/20/06, <b class="gmail_sendername">Alex Robar</b> <<a href="mailto:alex.robar@gmail.com">alex.robar@gmail.com</a>> wrote:<div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi Ralph,<br><br>Have you setup your PAP2 to allow the 729 codec? I believe you actually have to tell it that it's allowed to use that codec before it will work.<br><br>Cheers,<br>Alex<br><br><div><div><span class="e" id="q_10f0594a05f84372_1">
<span class="gmail_quote">
On 11/20/06, <b class="gmail_sendername">Ralph Liebessohn</b> <<a href="mailto:ralphliebessohn@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">ralphliebessohn@gmail.com</a>> wrote:
</span></span></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><span class="e" id="q_10f0594a05f84372_3">
Hi guys,<br>
<br>I've registered some g729 licenses, during register process everything worked fine.<br>
<br>
astk2*CLI> show g729<br>
0/0 encoders/decoders of 20 licensed channels are currently in use<br>
<br>But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk:<br>
<br>
Got SIP response 488 "Not Acceptable Here" back from <a href="http://192.168.10.126" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.10.126</a><br>
<br>And when I try to forward a call using IAX I got it:<br>
<br>
-- Executing Dial("SIP/teste1-081b23c8", "IAX2/192.168.5.12/6551|30|tT") in new stack<br>
-- Called <a href="http://192.168.5.12/6551" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.5.12/6551</a><br>
Nov 20 01:29:48 WARNING[8297]: chan_iax2.c:7099 socket_read: Call rejected by <a href="http://192.168.5.12" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.5.12</a>: Unable to negotiate codec
<br>
<br>Should asterisk translate to another codec when trying to make a new call with iax? Why can't asterisk make a call using g729 and sip?<br>
<br>Some configuration.<br><br>SIP.CONF<br>[teste]<br>
type=friend<br>
secret=blah<br>
host=dynamic<br>
username=teste<br>
context=from-spo<br>
disallow=all<br>
allow=g729<br>
allow=ulaw<br>
<br>
[teste1]<br>
type=friend<br>
secret=blah<br>
host=dynamic<br>
username=teste1<br>
context=from-spo<br>
disallow=all<br>
allow=g729<br>
allow=ulaw<br>
<br>EXTENSIONS.CONF<br>
exten => 1,1,Dial(SIP/teste1)<br>
exten => 0,1,Dial(SIP/teste)<br clear="all"><br>Thanks.<br><span><br>-- <br>Ralph Liebessohn<br>ICQ: 74835911<br>Skype: liebessohn
</span></span></div></blockquote></div><br>-- <br>Alex Robar<br><a href="mailto:alex.robar@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">alex.robar@gmail.com</a>
<br></blockquote></div><br><br>Hi Alex,<br><br>I set on Audio configuration to enable g729a, g729a as preferred codec and use only preferred codec. Is only that right?<br>With ulaw all calls work fine.<br clear="all"><br>
-- <br>Ralph Liebessohn<br>ICQ: 74835911<br>Skype: liebessohn