[Asterisk-Users] Re: How to prevent native bridging between SIP channels

[Wolf N. Paul]

Marc Storck <marc.storck at msnetworks.lu> 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