asterisk users - Feb 2005 - Polycom Auto-Answer and Call Transfers

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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> 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.