Thank you, the absolute codec string addition worked. My original dialstring was <action application="bridge" data="sofia/gateway/voip.ms/1$1"/>, BTW (where voip.ms 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? On Fri, Feb 4, 2011 at 12:07 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
try putting your dial string to your provider like this:
{absolute_codec_string=PCMU}sofia/foo/foo@bar.com
On Fri, Feb 4, 2011 at 10:45 AM, Marcin Wojtowicz <marcin321@gmail.com> wrote:
I have the following setup: voip.ms < (external profile) > freeswitch < (internal profile)> my devices. On my internal profile, the only inbound and outbound codec I allow is iLBC (I tried 20ms and 30ms), and on my external profile the only inbound and outbound codec I allow is PCMU. All my internal settings are compatible with the devices that I'm using, and everything works well when I call from outside in, and freeswitch now correctly transcodes from PCMU to iLBC. However, when I call from my internal devices to outside, the iLBC codec is offered on outbound leg B, not PCMU like I have on my external profile. Looking at wireshark voip call flow, my machine is sending iLBC to voip.ms and voip.ms is sending me PCMU RTP.
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