[Asterisk-Users] Starting RTP with Dial and MusicOnHold

Aaron Clauson aza at azaclauson.com
Thu Dec 15 07:49:51 MST 2005


> -----Original Message-----
> From: Elton Machado [mailto:elton.machado at gmail.com] 
> Sent: 15 December 2005 14:03
> To: aza at azaclauson.com; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
> 
> Why not to use r option in Dial(SIP/xyz,,r) to simulate the ring? 
>  
>  
> Regards, 
>  

Hi Elton,

Tried that one as well.

The Dial(,,r) command actually does the opposite of what I want. The r
option specifies that no audio, i.e. no RTP stream, should be passed until
the call is answered. This option will generate a SIP 180 Ringing response
on an incoming call but since in this case the Cerpack switch needs out of
band signalling any 180, 183 or other SIP repsonses are ignored for call
progress indication.

Thanks,

Aaron





More information about the asterisk-users mailing list