Thanks to the help of every one in this thread I have managed to get it working. The important thing I realized is it is necessary to stop Freeswitch whenever you change the settings and restart it again. I thought doing a reloadxml would get the changes to take, but that is not the case. I have documented my findings on my <http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura> devblog<http://devblog.brahmancreations.com/content/freeswitch-g729-pass-through-for-linksys-sipura> On 6 April 2010 08:10, Ognjen Seslija <oseslija@gmail.com> wrote:
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.
Since this is becoming a neverending saga, join us on IRC for help.
I have got to start using IRC more often, I have never really gotten into it, but I see that it is necessary when time is important. I prefer mailing lists though as the keep a better record of what has transpired for all to see. Thanks to all. My next question will be coming up shortly, this time on IRC first.
On Mon, Apr 5, 2010 at 11:02 PM, Frank Church <vfclists@googlemail.com> wrote:
I have changed it and this time I get this error
2010-04-05 21:55:22.640625 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode! 2010-04-05 21:55:22.640625 [ERR] switch_core_io.c:726 Codec G.729 decoder error!
When it is on 0.030 I get the garbled sound.
I guess the 0.020 is more compatible and I now have to set the pass through mode. Is there a parameter for that?
On 5 April 2010 21:47, Brian West <brian@freeswitch.org> wrote:
change it to 0.020
/b
On Apr 5, 2010, at 3:43 PM, Frank Church wrote:
I can't see a ptime in the Linksys settings, is that what is labelled as 'RTP Packet Size' on the SIP page?
It is the only thing that has the value 0.030
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