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Summary:ASTERISK-11404: rejected due to usage limit of 1
Reporter:Josue Conti (josueconti)Labels:
Date Opened:2008-02-08 08:43:44.000-0600Date Closed:2011-06-07 14:00:51
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Applications/app_queue
Versions:Frequency of
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Description:Hi all.
Friends would like to report the following error in the calls received for sip agents of a line call to call center:
The agent does not receive no call and asterisk reports the following one:
-- Accepting call from ' 1000 ' you the ' 19817 ' on channel 0/25, span 3 -- Executing Macro("Zap/87-1", "voicemail|sip/19817|19817|60|rTt") in new stack -- Executing Dial("Zap/87-1", "sip/19817|30tT") in new stack Feb 8 10:08:01 NOTICE[5941 ]: chan_sip.c:2273 to update_call_counter: Call you peer ' 19817 ' rejected due you usage limit of 1 -- Couldn't call 19817 == Everyone is busy/congested at this teams (0:0/0/0) -- Executing VoiceMail("Zap/87-1", "u19817") in new stack Feb 8 10:08:01 WARNING[5941 ]: app_voicemail.c:2461 leave_voicemail: In entry in voicemail config file will be ' 19817 ' -- Channel 0/9, span 3 got hangup request -- Stopped music on hold on Zap/71-1 -- User disconnected from queue 0800 while waiting to their turn == Spawn extension (sub0800, 5, 2) exited non-zero on ' Zap/71-1 ' Would like to know as to resolve this case?
Also to decide logoff automatic of the agents.

****** ADDITIONAL INFORMATION ******

S.O OpenSUSE 10.2
Asterisk-1.2.18
Server HP Proliant 380G5
Comments:By: Joshua C. Colp (jcolp) 2008-02-08 08:46:01.000-0600

Please seek help using normal methods (such as the asterisk-users mailing list or the #asterisk IRC channel) as we do not provide help on here, and we no longer accept issues related to 1.2.