[Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server

[Peter Svensson]

On Tue, 26 Apr 2005, raymond wrote:

> To my surprise, I change the Dial statement in extensions.conf from:
> 
> exten => _852.,1,Dial,SIP/123456${EXTEN}@192.168.11.194,r
> to:
> exten => _852.,1,Dial(SIP/123456${EXTEN}@192.168.11.194,20,r)
> I can hear ringback tone now.  I don't know why but it just works.

In the first line you passed th "r" in the argument reserved for the 
timeout value. Th options field in Dial is the third argument, not the 
second. So, you had a timeout of "r" seconds (invalid) and no ringback 
option.

Peter