Warren,

 

Thanks, I ended up doing that but it didn't change a thing. I mean, the
originating phone does not drop into a conference obviously, but the ringing
still goes on and on for 30 secs (my timeout).

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, August 08, 2011 2:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom and auto answer

 

On Sun, Aug 7, 2011 at 9:32 PM, Mike <l...@net-wall.com> wrote:

Hi,

[paging]

exten => s,1,Verbose(1,paging)
exten => s,n,Set(TIMEOUT(absolute)=30) ;to prevent call from being stuck
exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)
exten => s,n,Page(SIP/sipphone)

 





Try changing the Page() to a Dial() command and see if that makes a
difference.

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_____________________________________________________________________
-- 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

Reply via email to