Marc Storck <[EMAIL PROTECTED]> writes in reply to my question:
add
canreinvite=no
to the sip user definition blocks for the SIP provider and for the SIP ATA.
Regards,
Unfortunately, I already have this parameter in the sip user definitons, as well as
a "t" option in the Dial command, both of which, according to the article on
SIP Media Path in the Asterisk-Wiki, should prevent Asterisk from trying to take
itself out of the loop. But it still does :-(
On the other hand, the same article says that Asterisk decides whether or not to
take itself out of the media path depends on "many variables" -- I was hoping to
get some information on some of the _other_ variables, in addition to the
canreinvite=no and the "t"ransfer option to the Dial command.
Regards,
Wolf
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