[Asterisk-Users] RE: No incoming audio on incoming SIP calls

[John Howard]

Hi Steve, 

Sorry to hijack the thread, but I'm confused, when I add 'tr' to the end of
my dial strings to enable the transferring of that call internally, it
breaks asterisk's dial plan totally.  Calling any extension that has the tr
gives this error:

Aug  4 18:25:33 WARNING[17422]: app_dial.c:838 dial_exec: Invalid timeout
specified: 'tr'

I cant find many resources on this error, do I need to set up the
definitions of t and r (t for 'called extension' transferring, r to tell the
person the extension is ringing.)

Im using the take16.txt ## transfer patch from twister on the bug fix page
(number 2010) and the 03/08/04 checkout from the cvs.

The ## transfer doesn’t work either needless to say, sending the ## just
goes out over the audio stream as DTMF.

If you can point me at anything in the right direction it would be
appreciated.

Cheers,

jd

-----Original Message-----
t and T are for transfer, not timeout, case denotes which end can
transfer.
-- 
Steven Critchfield <critch at basesys.com>
 

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