<br><br><div class="gmail_quote">On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh <span dir="ltr"><<a href="mailto:prince@drishti-soft.com">prince@drishti-soft.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br><ol><li>Are Asterisk and Mittel in the same physical LAN.. or do they have a router between them?</li><li>Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data being sent to</li><li>Probable issues:-</li>
<ol><li>canreinvite is enabled when it should not be</li><li>Mitel is sending SDP with an incorrect RTP IP and/or port... You'll need to check 'sip debug' to see what RTP port is being sent</li></ol><li>From the 1/2 second audio, it seems that it could be due to one of these:-</li>
<ol><li>1/2 second is early media, and is being handled correctly at both Mitel and Asterisk. OR,</li><li>After 1/2 second, Asterisk and Mitel renogotiate for RTP payload type, and switch to a codec that is broken at either or both the locations</li>
<li>After 1/2 second, Asterisk and Mitel renogotiate for RTP IP/port<br></li></ol></ol>
<br>In case you are unable to debug with the above help, post these:-<br><ol><li>IPs of both Mitel and Asterisk<br></li><li>SIP dialog as text (sip debug output should do)</li><li>A few lines of RTP debug output</li></ol>
-- <br>Regards,<br>Prince Singh<br><br>Drishti-Soft Solutions Pvt Ltd<br><br></blockquote><div><br><br><br>Thank you for the feedback, 4.1 about early media led me to the answer! Your ideas and <a href="http://voip-info.org">voip-info.org</a> searching helped!<br>
<br>my extensions.conf was like this:<br>answer()<br>cut()<br>dial()<br><br>I changed it to:<br>cut()<br>dial()<br><br>Thanks again for your assistance!<br><br>-Thermal<br><br> <br></div></div><div style="visibility: hidden; display: inline;" id="avg_ls_inline_popup">
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