Re: [OpenSER-Users] Avoid Loops between Openser and Asterisk

2008/4/12, Adrian A adrianvoip@gmail.com:

If the INVITE comes in from Asterisk, OpenSER replies with 480 and lets Asterisk deal with sending call to voicemail (e.g. dialstatus = unavailable). This eliminates the loop because the INVITE does not come back to Asterisk.

Ok ok, I understand what you mean. But that is just useful for redirection to voicemail and so. For example, imagine you have Asterisk as PSTN gateway. Imagine Asterisk receives a call from PSTN and routes it to OpenSer, and OpenSer destination user has a redirection to a mobile number. Then OpenSer would route the invite back to Asterisk who will detect a loop (an spiral in fact, but Asterisk is buggy here).

Regards.