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We have polycom 501 phone. Do you know how to configure it to send back busy signal ?<br><br>> From: EWieling@nyigc.com<br>> To: asterisk-users@lists.digium.com<br>> Date: Mon, 2 May 2011 17:07:22 -0400<br>> Subject: Re: [asterisk-users] sip busy detect<br>> <br>> <br>> We always rely on our phones to send back a busy when busy. Is there a reason you can't do that?<br>> <br>> -----Original Message-----<br>> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of satish patel<br>> Sent: Monday, May 02, 2011 5:04 PM<br>> To: asterisk-users<br>> Subject: Re: [asterisk-users] sip busy detect<br>> <br>> <br>> Thanks for reply,<br>> <br>> I had tried to increase call-limit=2 or more also removed and in that case i am hearing ringing not detecting busy channel :(<br>> <br>> <br>> > From: EWieling@nyigc.com<br>> > To: asterisk-users@lists.digium.com<br>> > Date: Mon, 2 May 2011 16:59:10 -0400<br>> > Subject: Re: [asterisk-users] sip busy detect<br>> ><br>> > Remove your call-limit or increase your calllimit above your busy level<br>> ><br>> > -----Original Message-----<br>> > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of satish patel<br>> > Sent: Monday, May 02, 2011 4:56 PM<br>> > To: asterisk-users<br>> > Subject: [asterisk-users] sip busy detect<br>> ><br>> > Hi,<br>> ><br>> > I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf<br>> ><br>> > [7527]<br>> > type=friend<br>> > context=from-sip<br>> > host=dynamic<br>> > dtmfmode=rfc2833<br>> > callerid="Guest" <7527><br>> > mailbox=7527@default<br>> > nat=no<br>> > qualify=yes<br>> > cc_agent_policy=generic<br>> > cc_monitor_policy=generic<br>> > busylevel=1<br>> > limitonpeers=yes<br>> > call-limit=1<br>> ><br>> > when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ?<br>> ><br>> > [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1<br>> > -- Couldn't call 7527<br>> > -- Called 7527<br>> > == Everyone is busy/congested at this time (1:0/1/0)<br>> > -- Executing [s@macro-stdexten:2] Goto("SIP/7604-00000006", "s-CONGESTION,1") in new stack<br>> > -- Goto (macro-stdexten,s-CONGESTION,1)<br>> ><br>> ><br>> ><br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> <br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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