There isn't one. Both call legs need to agree on G729. Try to set only G729 in inbound/outbound codec-prefs in sip profile or use disable-transcoding.<br><br>Since this is becoming a neverending saga, join us on IRC for help.<br>
<br><div class="gmail_quote">On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <span dir="ltr"><<a href="mailto:vfclists@googlemail.com">vfclists@googlemail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I have changed it and this time I get this error<br>
<br>
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only<br>
usable in passthrough mode!<br>
2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error!<br>
<br>
When it is on 0.030 I get the garbled sound.<br>
<br>
I guess the 0.020 is more compatible and I now have to set the pass<br>
through mode. Is there a parameter for that?<br>
<div><div></div><div class="h5"><br>
<br>
On 5 April 2010 21:47, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<br>
> change it to 0.020<br>
><br>
> /b<br>
><br>
> On Apr 5, 2010, at 3:43 PM, Frank Church wrote:<br>
><br>
>> I can't see a ptime in the Linksys settings, is that what is labelled<br>
>> as 'RTP Packet Size' on the SIP page?<br>
>><br>
>> It is the only thing that has the value 0.030<br>
><br>
><br>
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<br>
</div></div>--<br>
<div class="im">Frank Church<br>
<br>
=======================<br>
<a href="http://devblog.brahmancreations.com" target="_blank">http://devblog.brahmancreations.com</a><br>
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