<div>That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk.<br></div><div><br></div><div>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/</div>
<div><br></div><div>I don't get it!...</div><div><br></div><div>More ideas?</div><div><br></div><div>Thanks,</div><div>Ricardo.</div><div><br></div><div><br></div><br><div class="gmail_quote">On Wed, May 9, 2012 at 3:32 PM, A J Stiles <span dir="ltr"><<a href="mailto:asterisk_list@earthshod.co.uk" target="_blank">asterisk_list@earthshod.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">On Wednesday 09 May 2012, Ricardo Carvalho wrote:<br>
<br>
> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible<br>
> codecs, not accepting this offer!<br>
><br>
> Any help?<br>
<br>
</div>Are you sure you compiled all the codecs you need?<br>
<br>
What happens if you run `make menuselect` in both the 1.4 source tree and in<br>
the 1.8 source tree, "side-by-side" in tabs of the same terminal window? You<br>
need at least GSM, A-law and micro-law.<br>
<br>
(The above is my preferred method of building a configuration like an existing<br>
one. No doubt someone will weigh in with a better way of doing it.)<br>
<br>
--<br>
AJS<br>
<br>
Answers come *after* questions.<br>
<br>
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</blockquote></div><br>