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

Benjamin Jacob ben4asterisk at yahoo.com
Tue Apr 22 06:54:50 CDT 2008


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

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


       
---------------------------------
Be a better friend, newshound, and know-it-all with Yahoo! Mobile.  Try it now.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080422/a2753a4d/attachment-0001.htm 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: reInvite
Type: application/octet-stream
Size: 24529 bytes
Desc: 1957794313-reInvite
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080422/a2753a4d/attachment-0001.obj 


More information about the asterisk-users mailing list