I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER, so that Phone A is connected with the ringing Phone C. Asterisk sends an UPDATE to Phone-C to update the connected line information. Now the user on…
The DIALSTATUS channel variable is created when you attempt to connect to another device or endpoint and bridge the call with the Dial Application. It contains the status of the call reflected in one of the following values:
- DONTCALL - For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the 'Go Away' script.
- TORTURE - For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the 'torture' script.
i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server
the call flow
IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE
5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP
phone (5000) my case is:
1/ IP phone(5000) --->Opensips
2/ IVR number : 1001
3/ IP Phone calls 1001 to opensips --> asterisk, ASterisk will play IVR
4/ IP phone press 1, asterisk will Dial(SIP/to_opensips/5001,20)
5/ there are some cases when asterisk send call back to opensips