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

[Noah Miller]

> 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.

Instead of trying to hack something out of the SIP headers, you could have the dialplan take care of this.  Maybe something like the solution indicated in the WIKI for the Polycom auto-answer - dial 8 before an extension to auto-answer, otherwise the extension will ring. Something like:

exten => _8XXX,1,SetVar(ALERT_INFO=A-A)
exten => _8XXX,2,Dial(SIP/${EXTEN:1},20)
exten => _8XXX,3,Hangup

I know this isn't exactly what you were looking for because it means the users have to remember to dial a different number to intercom than to transfer with ringing, but it works.  If you are under a time crunch you could do this immediately, and figure out the SIP header hacking as time permits.