[Asterisk-Users] Problems with "r" modifier in Dial - does not work in SIP channels?

[John Todd]

I can't seem to get the "r" modifier to work on inbound SIP calls. 
The way I understood this to work is that the channel would be 
answered, and a ring "tone" would be played to the channel.  This is 
not very friendly in that it doesn't honor connection supervision 
rules, but... who cares?  There are some instances where it may be in 
my interests to get a "ringing" sound played to the caller.  As an 
example, I have a service that hands me calls and does not do correct 
connection supervision tone playback, and thus the caller doesn't 
hear ringtones.  This is Bad, so I figured I'd just have Asterisk 
immediately answer the line and play ringtones into the channel until 
I answered.  But that method doesn't seem to work.

I've tried all of the combinations below.  At best, I'll hear one 
"ring" tone, and then silence until the Dial times out.  ${PHONE} is 
a SIP/xxxx extension, and the calls to these routines are also coming 
in via SIP.  Voice works fine on the channels, once answered.

Does anyone know if there is something wrong with the "r" modifier on 
SIP Dial application calls, or have you had experience doing this a 
better way?


None of these methods work:

exten => 1234,1,Dial(${PHONE1},25,r)


exten => 1234,1,Answer
exten => 1234,2,Dial(${PHONE1},25,r)


exten => 1234,1,Answer
exten => 1234,2,Ringing
exten => 1234,3,Dial(${PHONE1},25,r)


exten => 1234,1,Ringing
exten => 1234,2,Dial(${PHONE1},25,r)


exten => 1234,1,Ringing
exten => 1234,2,Dial(${PHONE1},25)



JT