so you need log from ‘sip debug’ ?
here is full log, cancel is here:
[Feb 28 22:09:37] CANCEL sip:64668@domainxxx SIP/2.0
and here is a small log:
[Feb 28 22:21:41] -- Registered SIP '8003' at 192.168.0.168:1683
[Feb 28 22:21:57] == Using SIP VIDEO CoS mark 6
[Feb 28 22:21:57] == Using SIP RTP CoS mark 5
[Feb 28 22:21:57] -- Executing [8100@default:1] Verbose("SIP/8002-00000000", "2,Call desk phone and mobile but with delay") in new stack
[Feb 28 22:21:57] == Call desk phone and mobile but with delay
[Feb 28 22:21:57] -- Executing [8100@default:2] Dial("SIP/8002-00000000", "Local/indoor1@extensions&Local/mobile1@extensions,30") in new stack
[Feb 28 22:21:57] -- Called Local/indoor1@extensions
[Feb 28 22:21:57] -- Executing [indoor1@extensions:1] Verbose("Local/indoor1@extensions-00000000;2", "2,Dialing indoor station") in new stack
[Feb 28 22:21:57] == Dialing indoor station
[Feb 28 22:21:57] -- Executing [indoor1@extensions:2] Dial("Local/indoor1@extensions-00000000;2", "SIP/8003") in new stack
[Feb 28 22:21:57] -- Called Local/mobile1@extensions
[Feb 28 22:21:57] -- Executing [mobile1@extensions:1] Verbose("Local/mobile1@extensions-00000001;2", "2,Dialing mobile with waiting time") in new stack
[Feb 28 22:21:57] == Dialing mobile with waiting time
[Feb 28 22:21:57] -- Executing [mobile1@extensions:2] Wait("Local/mobile1@extensions-00000001;2", "3") in new stack
[Feb 28 22:21:57] == Using SIP VIDEO CoS mark 6
[Feb 28 22:21:57] == Using SIP RTP CoS mark 5
[Feb 28 22:21:57] -- Called SIP/8003
[Feb 28 22:21:57] -- Local/indoor1@extensions-00000000;1 connected line has changed. Saving it until answer for SIP/8002-00000000
[Feb 28 22:21:58] -- SIP/8003-00000001 is making progress passing it to Local/indoor1@extensions-00000000;2
[Feb 28 22:21:58] -- Local/indoor1@extensions-00000000;1 is making progress passing it to SIP/8002-00000000
[Feb 28 22:22:00] -- Got SIP response 603 "Decline" back from 192.168.0.168:1683
[Feb 28 22:22:00] -- SIP/8003-00000001 is busy
[Feb 28 22:22:00] == Everyone is busy/congested at this time (1:1/0/0)
[Feb 28 22:22:00] -- Auto fallthrough, channel 'Local/indoor1@extensions-00000000;2' status is 'BUSY'
[Feb 28 22:22:00] -- Local/indoor1@extensions-00000000;1 is busy
[Feb 28 22:22:00] -- Executing [mobile1@extensions:3] Set("Local/mobile1@extensions-00000001;2", "CALLERID(num)=64668") in new stack
[Feb 28 22:22:00] -- Executing [mobile1@extensions:4] Dial("Local/mobile1@extensions-00000001;2", "SIP/64668@trunk") in new stack
[Feb 28 22:22:00] == Using SIP VIDEO CoS mark 6
[Feb 28 22:22:00] == Using SIP RTP CoS mark 5
[Feb 28 22:22:00] -- Called SIP/64668@trunk
[Feb 28 22:22:01] -- SIP/trunk-00000002 is ringing
[Feb 28 22:22:01] -- Local/mobile1@extensions-00000001;1 is ringing
[Feb 28 22:22:10] == Spawn extension (extensions, mobile1, 4) exited non-zero on 'Local/mobile1@extensions-00000001;2'
[Feb 28 22:22:10] == Spawn extension (default, 8100, 2) exited non-zero on 'SIP/8002-00000000'
you see i did a decline there, and its still calling the mobile one
The long trace shows a successful CANCEL across all three lines. If one is still ringing, the problem is downstream.
I don’t think that Asterisk really understands “…everywhere” type failures. It uses ISDN codes internally, and the official translation for busy here is the same as busy everywhere, so it can’t distinguish in that case, and may well take the same position on decline,
This is with dial, but that would be even more true of a ringall queue; you wouldn’t want an agent who was too busy throwing the caller out completely.
Hmm, so not possible then?
would it help if i use a queue config instead?
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