Thanks for reply Leandro.<div><br></div><div>We have installed g279 codec in Asterisk box.Even if not so, there is no problem outgoing (from Asterisk to CCM) calls. But after i searched the issue, i figured out that CCM 4.x does not let g729 codec to pass through over SIP trunk. This is limited only in CCM. If we changed codec g729 into g711u (ulaw) then communication over SIP trunk go on perfectly. </div>
<div><br></div><div>Because of CCM does not inject any packets encoded g729 over SIP trunk, i am not able to debug it. But i have tried that i am able to force my SIP phone suscribed Asterisk box to use g729 codec and get work successfully.</div>
<div><br><div class="gmail_quote">2013/1/17 Leandro Dardini <span dir="ltr"><<a href="mailto:ldardini@gmail.com" target="_blank">ldardini@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="HOEnZb"><div class="h5">2013/1/17 Onur Cem Çelebi <span dir="ltr"><<a href="mailto:occelebi@gmail.com" target="_blank">occelebi@gmail.com</a>></span><br></div></div><div class="gmail_quote"><div><div class="h5">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br><br>My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13)
and solution seems to have H323 trunk between CCM and Asterisk for
using g729 codec. The post was written in 2006. Is there any better
solution since that time ? Thanks for reading.<br> <br>link : <a href="https://supportforums.cisco.com/message/1072037" target="_blank">g279 codec over SIP Trunk between CCM and Asterisk</a><br><br><br></blockquote><div>
<br>
</div></div></div><div>Have you checked if the problem is the license? Asterisk doesn't have a free encoder/decoder for g729, only pass through is available. Try to debug the SIP call to see if the capabilities don't match or just buy a $10 license from Digium (1 concurrent call).</div>
<span class="HOEnZb"><font color="#888888">
<div><br></div><div>Leandro </div></font></span></div>
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