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

[SIP]

Russell Bryant wrote:
> 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.
>
>   
It's a rather simple config, really.

Peer:

[vitel-termination]
type=peer
host=outbound1.vitelity.net
username=myuser
fromuser=myuser
trustrpid=yes
sendrpid=yes
secret=mysecret
allow=all
canreinvite=yes



Call file:

Channel: SIP/1XXXXXXXXXX at vitel-termination
MaxRetries: 1
RetryTime: 10
WaitTime: 30
Application: AGI
Data: /usr/local/click-to-call/dialtest.pl|1YYYYYYYYYY
Callerid: XXXXXXXXXX



The dialtest.pl just issues a dial to the 1YYYYYYYYYY number after 
Asterisk rings the 1XXXXXXXXXX number. Very basic, simple click-to-call 
stuff.

You think maybe it's an issue with transcoding? I haven't tried forcing 
a single codec to see, but I'm pretty sure that Vitelity does g711 and 
g729. As we don't have any g729 licenses installed, would it not simply 
ignore those (not sure how Asterisk negotiates codecs in a situation 
like that, to be honest. It's always seemed like a bit too much 
jiggerypokery for my tastes) ?


N.