[Asterisk-Users] Polycom Auto-Answer and Call Transfers

[Jared Armstrong]

I have my * and polycom system setup to do Auto-Answer for internal
SIP/Staff calls, and I am running into an issue with this and the
polycom call transfer feature. * is seeing a new call come through from
the polycom and is then transferring the call over. I need to know if
there is some way I can grab a message from the SIP header or something
to determine if I should not set the ALERT_INFO tag to A-A. I would
greatly appreciate it if someone could help me out with this, I need to
have this resolved by Monday.

 

Thanks,

 

Jared Armstrong

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