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

[Tobias Jönsson]

On Thu, 5 Aug 2004, John Howard wrote:

> 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'

Probably you put the tr options in the timeout field. Forgot a comma in
the dial command? As said on
<http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial> the
syntax is Dial(type/identifier,timeout,options,URL) so if you want to dial
SIP/kalle with no timeout with options tr you should put
Dial(SIP/kalle,,tr) in your extensions.conf.

Regards,
Tobias Jönsson, Lund, Sweden