[asterisk-users] Getting Asterisk out of the RTP media path

[SIP]

SIP wrote:
> When calling from our SIP proxy through Asterisk to the PSTN provider, 
> we support reINVITES which tend to work seamlessly.
>
> However, when creating a call file which essentially connects a call 
> from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP 
> media path. I understand that this is sort of the idea behind a bridged 
> channel, but is there any way to avoid it? Is there any way to say 
> "Connect this number and this number and then get out of the way,"  or 
> is this a design limitation?
>
> N.
>
>   

No ideas on this one? I've tried everything I can think of and then some 
and still can't get Asterisk out of the media path. I can do it if I 
don't originate the call with Asterisk, but only use Asterisk to connect 
one leg of the call, but if I use Asterisk to connect both legs, no luck.

Going about this the wrong way?


N.