OK, it's been a while since I drank from the pool of wisdom hear on the list.
After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be dialed). The purpose of this is to allow the inbound calls to come in, then if the operator has a need, they transfer the call to a pre-defined extension which lives in the limited context defined in sip.conf. This has worked for some time to restrict outbound calling and where calls can be transferred to. Now I would like to open up the numbers the inbound calls can be transferred to. So, easy enough I thought and I went on my merry way adding the regular patterns to the context such as NXXNXXXXXX and so on. Hooray, now the operators can transfer anywhere. New Problem, now operators can pick up the previous inbound only line and dial out to anything that matches the patterns I have defined in the context for their extension in sip.conf. What I really need to make work here is Attended-Transfer since that is what is desired by those using the system. It seems that any variables I try to set on the way in don't carry through too well during an attended transfer. Basically, I need the ability to know for sure at the point the call ends up in the outbound context (defined in sip.conf) if the call is actually a transfer from an inbound call or if it's a direct dial outbound call with no incoming call attached. If I can figure out how to know this for sure, I can just do a GoToIf type of thing in the outbound context that just kills the call if there is no proof that it's a transfer. I hope this makes sense, please let me know if more info is needed. Running Asterisk 1.8.8.0. A huge thanks in advance to the list for any help with this, it's driving me batty. Regards, Todd R.
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
