On 5/9/2012 11:56 AM, Ricardo Carvalho wrote:
That's weird, because it's negotiated with success the codec ulaw for
outbound calls through the same SIP trunk.
My guess is the incoming call is not being matched with the peer you are
expecting. Do a sip debug and watch the output to see what peer is
being selected.
Andres
Besides, ulaw and alaw shows up when i do "core show codecs audio" in
the asterisk CLI, and there exists both codec_ulaw.so and
codec_alaw.so modules under the path /usr/lib/asterisk/modules/
I don't get it!...
More ideas?
Thanks,
Ricardo.
On Wed, May 9, 2012 at 3:32 PM, A J Stiles
<[email protected] <mailto:[email protected]>>
wrote:
On Wednesday 09 May 2012, Ricardo Carvalho wrote:
> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No
compatible
> codecs, not accepting this offer!
>
> Any help?
Are you sure you compiled all the codecs you need?
What happens if you run `make menuselect` in both the 1.4 source
tree and in
the 1.8 source tree, "side-by-side" in tabs of the same terminal
window? You
need at least GSM, A-law and micro-law.
(The above is my preferred method of building a configuration like
an existing
one. No doubt someone will weigh in with a better way of doing it.)
--
AJS
Answers come *after* questions.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
