asterisk users - Aug 2008 - Getting Asterisk out of the RTP media path

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.

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.