[Asterisk-Users] extensions problem (SIP)

Olle E. Johansson oej at edvina.net
Mon Mar 15 09:00:42 MST 2004

Jon Lawrence wrote:
> Hi,
> I've got 2 x100p's installed in my system.
> Both execute the same incoming contexts as follows:
> [inboundA]
> include => dialjon
> [inboundB]
> include => dialjon|09:00-16:30|Mon-Fri|*|*
> [dialjon]
> exten => s,1,answer
> exten => s,2,Dial(SIP/2000,15)
> exten => s,3,Playback(noone)
> exten => s,103,Goto(onphone,s,1)
> <snip>
> Am I right in saying:
> if no one answers at ext 2000 then s,3 is executed.
> if ext 2000 is busy  then 103 is executed.
> If so then sometihng is wrong. If I'm already on a call, I want 103 to be 
> executed however, this isn't happening. If a new call comes in (whilst I'm 
> talking on ext 2000) I here it ringing on my handset.

It depends on your SIP device. Asterisk places the call to your SIP device regardless,
since by SIP protocol design the UA is not a "slave", it is free. So the SIP ua must
answer "busy" for Asterisk to understand that you're busy. If not, the call is placed
to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can
limit the number of concurrent calls.

There's some code in Asterisk chan_sip.c to limit the number of calls placed to
a SIP phone, but right now it's not working at all.


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