Hey, I have one pstn line thats split into 2, one goes to a regular phone and the other one goes to asterisk. When someone calls I want a SIP phone to ring along with the regular phone. I have the context for the Zap interface set correctly and everything I even see the incomming call from Zap and the outgoing call on SIP being initialized... the SIP phone isnt ringing.
The idea is that the first phone (SIP or reg pstn phone) to pick up the line gets the call...
here is my extensions.conf
[from-sip]
exten => 200,1,Dial(SIP/200,300)
exten => _.,1,Dial(Zap/1/${EXTEN})
exten => h,1,Hangup
[from-pstn]
exten => s,1,Dial(SIP/200,300)
Dialing 200 rings the SIP phone but incomming pstn lines dont eventho I see the Dial(SIP/200|300) being executed... i just dont see it riniging for some reason. Im wondering if that has anything to do with not having Answer() in there...
Tom
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I dnt think this is related to asterisk server (I may be wrong).
The scenario u said is equivalent to 2 phones on a single line ( Intecom). If u want first asterisk shud ring and u want to pick up call from another phone. THen u need to use DISA() function of asterisk in which with proper configuration of TDM400P cards u canachieve it.
No what the author I think is wanting is a fairly standard configuration and is what I use at home aswell.
An incoming call is detect and asterisk starts ringing the SIP phone. At that point both phones are ringing (2nd phone on analogue line and SIP phone).
If the SIP phone is picked up then the analogue line is answered and the phone connected to the line. If the analogue phone is answered then after a couple of rings asterisk will see the line has stopped ringing and and the SIP phone will stop.
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by: grbladesPosted on 2006-07-14 at 01:52:46ID: 17106602
That looks fine. Can you post your zapata.conf file.