Thank you, the absolute codec string addition worked. My original dialstring was <action application="bridge" data="sofia/gateway/<a href="http://voip.ms/1$1">voip.ms/1$1</a>"/>, BTW (where <a href="http://voip.ms">voip.ms</a> is imported by the external profile). I just have one more unrelated question about voice sample times. If one leg is using 20ms and the other 30ms, does freeswitch introduce a buffer and if so, how much delay is it?<br>
<br><div class="gmail_quote">On Fri, Feb 4, 2011 at 12:07 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.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;">
try putting your dial string to your provider like this:<br>
<br>
{absolute_codec_string=PCMU}sofia/foo/<a href="mailto:foo@bar.com">foo@bar.com</a><br>
<div><div></div><div class="h5"><br>
<br>
On Fri, Feb 4, 2011 at 10:45 AM, Marcin Wojtowicz <<a href="mailto:marcin321@gmail.com">marcin321@gmail.com</a>> wrote:<br>
> I have the following setup:<br>
> <a href="http://voip.ms" target="_blank">voip.ms</a> < (external profile) > freeswitch < (internal profile)> my devices.<br>
> On my internal profile, the only inbound and outbound codec I allow is iLBC<br>
> (I tried 20ms and 30ms), and on my external profile the only inbound and<br>
> outbound codec I allow is PCMU. All my internal settings are compatible with<br>
> the devices that I'm using, and everything works well when I call from<br>
> outside in, and freeswitch now correctly transcodes from PCMU to iLBC.<br>
> However, when I call from my internal devices to outside, the iLBC codec is<br>
> offered on outbound leg B, not PCMU like I have on my external profile.<br>
> Looking at wireshark voip call flow, my machine is sending iLBC to <a href="http://voip.ms" target="_blank">voip.ms</a><br>
> and <a href="http://voip.ms" target="_blank">voip.ms</a> is sending me PCMU RTP.<br>
><br>
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