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.
On Fri, Apr 11, 2008 at 12:33 AM, Iñaki Baz Castillo ibc@in.ilimit.es wrote:
El Thursday 10 April 2008 23:00:28 Adrian A escribió:
Here is how I solve this.
In Asterisk, I add a SIP header just before sending it to OpenSER using
the
function: SipAddHeader(P-hint: From Asterisk).
Then, in OpenSER I have the following:
# check to see if user is online if (!lookup("location")) { # SIP from Asterisk or not a call attempt if (search("(P-hint):.From Asterisk") ||!(method==INVITE)) { sl_send_reply("480","User Temporarily
Unavailable");
exit; } else { route(2); exit; }; };
route[2] { # Let Asterisk deal with voicemail/forward sl_send_reply("181", "Forward to Voicemail"); prefix("vm"); setflag(9); # forward to Asterisk route(1); }
I can't understand why that should work, in fact Asterisk is receiving the same INVITE he sent (well, RURI modified but the Asterisk bug is that it doesn't recogniza a spiral).
-- Iñaki Baz Castillo ibc@in.ilimit.es _______________________________________________ Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users