if i call from pstn-phone a number (010-1234 2000) then asterisk gets this call and send a sip-INVITE to SER-SIP-Proxy to firstname.lastname@example.org
This working good: the SIP-soft-phone is ringing.
If after 10 seconds nobody answer the call at email@example.com, then SER-Proxy redirect the call to a local pstn-phone with number 1000. SER rewrite the host:port with Asterisk location. But the pstn-phone with number 1000 doesnt ringing…
But is it working if i make the call from a voip-phone to firstname.lastname@example.org, SER redirects the call after 10 seconds to pstn-phone 1000. And it is ringing.
Any tipps how i can fix or debuging this?
works: voip (extern) -> my SER-Proxy -> redirect to asterisk -> pstn-phone (is ringing)
doesnt work: pstn (extern) -> asterisk -> my SER-Proxy -> redirect to asterisk -> pstn-phone (doesn ringing)