On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh <pri...@drishti-soft.com>wrote:
> > > 1. Are Asterisk and Mittel in the same physical LAN.. or do they have a > router between them? > 2. Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data > being sent to > 3. Probable issues:- > 1. canreinvite is enabled when it should not be > 2. 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 > 4. From the 1/2 second audio, it seems that it could be due to one of > these:- > 1. 1/2 second is early media, and is being handled correctly at both > Mitel and Asterisk. OR, > 2. 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 > 3. After 1/2 second, Asterisk and Mitel renogotiate for RTP IP/port > > > In case you are unable to debug with the above help, post these:- > > 1. IPs of both Mitel and Asterisk > 2. SIP dialog as text (sip debug output should do) > 3. A few lines of RTP debug output > > -- > Regards, > Prince Singh > > Drishti-Soft Solutions Pvt Ltd > > Thank you for the feedback, 4.1 about early media led me to the answer! Your ideas and voip-info.org searching helped! my extensions.conf was like this: answer() cut() dial() I changed it to: cut() dial() Thanks again for your assistance! -Thermal
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