Pstn -> asterisk -> sip(ser) -> asterisk(redirect)


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
This working good: the SIP-soft-phone is ringing.

If after 10 seconds nobody answer the call at, 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, 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)

i have now logged with ethereal and got this from Asterisk, after SER send an Invite to Asterisk:
SIP/2.0 482 Loop Detected

please can somebody tell me if it is possible to route a call back from asterisk to asterisk


On the loop detected issue you may want to use local channel LOCAL/${exten} to avoid the loop.