Great! let me try.. We have same extension configured on two line. is this option will allow call transfer and two way conference ?
See this thread http://forums.digium.com/viewtopic.php?t=3716 -S > From: [email protected] > To: [email protected] > Date: Mon, 2 May 2011 17:18:02 -0400 > Subject: Re: [asterisk-users] sip busy detect > > > We use the following in the Polycom config files. > > <call > call.callsPerLineKey="1" > /> > > This will allow one call per line key on the phone, when calls are on all the > line keys, the phone will return a busy. This will vary slightly if you use > a different registration for each line key, etc. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of satish patel > Sent: Monday, May 02, 2011 5:14 PM > To: asterisk-users > Subject: Re: [asterisk-users] sip busy detect > > We have polycom 501 phone. Do you know how to configure it to send back busy > signal ? > > > From: [email protected] > > To: [email protected] > > Date: Mon, 2 May 2011 17:07:22 -0400 > > Subject: Re: [asterisk-users] sip busy detect > > > > > > We always rely on our phones to send back a busy when busy. Is there a > > reason you can't do that? > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
