[asterisk-users] SIP. Call-limit dialstatus

[I.Pavlov]

Hi all. We have a problem with correct dialstatus and cdr(disposition) when
using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and
CDR(disposition)='NO ANSWER'

 

    -- Executing [0014 at sub_pbxdialco:49] Dial("SIP/1295-000001f8",
"SIP/0014,12,tTkK") in new stack

  == Using SIP RTP CoS mark 5

[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1

    -- Couldn't call 0014

  == Everyone is busy/congested at this time (0:0/0/0)

    -- Executing [0014 at sub_pbxdialco:50] NoOp("SIP/1295-000001f8",
"CHANUNAVAIL") in new stack

 

I think that isn't correct. Is it possible to change dialstatus and
CDR(disposition) to BUSY-value when call-limit reached?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130703/abfb6233/attachment.htm>