This is the big problem with using Asterisk for SIP. With Asteriskhttp://www.voip-info.org/wiki-Asterisk+sip+reinvite
the audio data between two SIP extensions has to actualy go into
then out of the Asterisk box. This does not scale well to
thousands of users like in a university campus or a comercial
SIP service.
As I understand this Asterisk sets up the call with itself as endpoints, then moves the stream tobypass the PBX and go directly with a SIP reinvite. Some clients does not support this, and with those you have to configure asterisk to stay in the media path for this client with canreinvite=no in SIP.conf.
/O
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