My guess is the problem I am describing is a simple fix:
I have managed to configure a pretty complicated lab environment and have most things working quite well overall. I have Exchange 2010 for voicemail, MS Lync for the ‘internal’ pbx and softphones plus a SIP trunk pointing to an AsteriskNOW server I built, that in turn uses Skype for Asterisk for PSTN access in and out - all of which is creating a very fun experience!
I am having a few issues related to calls being dropped if they go unanswered and the call does not ring long enough for a remote party’s voice mail to kick in.
Here’s the scenario:
Outbound call: Dial out from Lync, call gets passed to SIP trunk, AsteriskNOW receives SIP from Lync and then calls out to Skype using my BCP account without issue. If the recipient on the PSTN picks up before approx. three rings, the call completes successfully without issue. IF the recipient does not answer within approx. three rings - I show AsteriskNOW dropping the call and transferring to extension h for the hangup.
Incoming calls work fine and if I don’t answer the call it successfully gets picked up by Exchange and can leave a voice mail.
I have a very basic sip.conf/extensions.conf/chan_skype.conf but I imagine it has something to do with my dial plan on Asterisk but am not entirely sure where to start on this one.
Does anyone have any recommendations for a problem like this?