Hi, 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 [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [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 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-stdexten:2] Goto("SIP/7604-00000006", "s-CONGESTION,1") in new stack -- Goto (macro-stdexten,s-CONGESTION,1) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/283e20ce/attachment.htm>
Remove your call-limit or increase your calllimit above your busy level -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel Sent: Monday, May 02, 2011 4:56 PM To: asterisk-users Subject: [asterisk-users] sip busy detect Hi, 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 [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [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 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-stdexten:2] Goto("SIP/7604-00000006", "s-CONGESTION,1") in new stack -- Goto (macro-stdexten,s-CONGESTION,1)
Apparently Analagous Threads
- Busy level in Asterisk 11
- Asterisk QSIG doesnt send the calling name to Nortel CS1000
- Restart asterisk destroy all registered SIP peers
- DEVICE_STATE() and Asterisk 1.6.0.10
- Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration