I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone.
- If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup.
As system administrator, monitoring the continuity of services is vital. Today I would like to highlight some tools that could come in handy for VoIP monitoring.
I have a some non standard problem when I register my asterisk into "My SIP Provider" . The trouble is: my asterisk stay behind router with port forwarding, who have Public IP (220.127.116.11 - for example), asterisk have a private IP (192.168.1.2)
From "My SIP Provider" cabinet I see: , but I need from (example):
From wich trukes in linux or asterisk technology I need? Can you help?
-- Best regards, Nikolay G. Petrov!