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

[Russell Bryant]

On Aug 11, 2008, at 12:04 PM, SIP wrote:

> 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?
>>
> 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?


Asterisk will re-INVITE the media away from itself as long as it  
doesn't have a reason to need access to the media.  For example, if  
you've enabled call recording, then clearly Asterisk needs access to  
the media.  Other reasons include enabling features controlled via  
DTMF when the DTMF follows the media path.

Nobody can help any further without seeing the details of your  
configuration.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.