[asterisk-users] re-invite (bypass asterisk) post call establishment

[Benjamin Jacob]

Hi again,
I tried this again, but the reInvite happens immediately after the 200 OK/ACK. And then the D() specified DTMF is sent.

Attached is the SIP trace for the calls.
I call (from Asterisk) - 0119198807xxxxx 
After connect, I dial - 31927xxxxx.
This number 31927xxxxx is the conference bridge and I need to send DTMF (the bridge PIN) to it after connection. But alas, the reinvite happens before the D() is executed.
The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. 

cheers
- Ben.




Steve Davies <davies147 at gmail.com> wrote: 2008/4/22 Benjamin Jacob :
[snip]
>
> So, my question : once the SDPs are exchanged, what will happen to the DTMFs
> sent by Asterisk using sendDTMF or the D option in dial.
>
[snip]

As far as I can tell, the D() option will be executed before the
re-invite takes place, so Asterisk will still be in-line. I believe
that the dial is not considered "complete/connected" until the D() is
finished.

Cheers,
Steve

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