Hola ya he pasado anteriormente este error sin recibir respuesta.
Se produce en issabel 4 con asterisk 11 con todo acutualizado.
Al entrar una llamada procedente de un SIP TRUN y pasar esta a trabes de un IVR a una extensión, al colgar la llamada en la extensión se produce el hangupcall:
-- Executing [h@macro-dial-one:1] Macro("SIP/Sarenet-00000000", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/Sarenet-00000000", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,20)
-- Executing [s@macro-hangupcall:20] NoOp("SIP/Sarenet-00000000", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:21] GotoIf("SIP/Sarenet-00000000", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,39)
-- Executing [s@macro-hangupcall:39] NoOp("SIP/Sarenet-00000000", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:40] GotoIf("SIP/Sarenet-00000000", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] NoOp("SIP/Sarenet-00000000", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:46] GotoIf("SIP/Sarenet-00000000", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,52)
-- Executing [s@macro-hangupcall:52] NoOp("SIP/Sarenet-00000000", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:53] GotoIf("SIP/Sarenet-00000000", "1?noautomon3") in new stack
-- Goto (macro-hangupcall,s,59)
-- Executing [s@macro-hangupcall:59] NoOp("SIP/Sarenet-00000000", "MIXMONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:60] GotoIf("SIP/Sarenet-00000000", "1?noautomon4") in new stack
-- Goto (macro-hangupcall,s,62)
-- Executing [s@macro-hangupcall:62] NoOp("SIP/Sarenet-00000000", "ONETOUCH_RECFILE=") in new stack
-- Executing [s@macro-hangupcall:63] NoOp("SIP/Sarenet-00000000", "CDR recordingfile set to: ") in new stack
-- Executing [s@macro-hangupcall:64] GotoIf("SIP/Sarenet-00000000", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,67)
-- Executing [s@macro-hangupcall:67] GotoIf("SIP/Sarenet-00000000", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,70)
-- Executing [s@macro-hangupcall:70] GotoIf("SIP/Sarenet-00000000", "1?theend") in new stack
-- Goto (macro-hangupcall,s,72)
-- Executing [s@macro-hangupcall:72] AGI("SIP/Sarenet-00000000", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/Sarenet-00000000>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:73] Hangup("SIP/Sarenet-00000000", "") in new stack
== Spawn extension (macro-hangupcall, s, 73) exited non-zero on 'SIP/Sarenet-00000000' in macro 'hangupcall'
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/Sarenet-00000000'
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/Sarenet-00000000' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/Sarenet-00000000' in macro 'exten-vm'
== Spawn extension (from-did-direct, 8115, 2) exited non-zero on 'SIP/Sarenet-00000000'
Really destroying SIP dialog '1mmebggb3g01i0eafbki1iax442xd1fg@SoftX3000' Method: ACK
Pero la llamada no se termina hasta que el que ha llamado cuelga, en ese momento se recibe:
<--- SIP read from UDP:194.30.6.32:5060 --->
BYE sip:912900324@2.137.182.204:5060 SIP/2.0
Via: SIP/2.0/UDP 194.30.6.32;branch=z9hG4bK8094.16fe953123d66f6213332f91515e3e8f.0
Via: SIP/2.0/UDP 194.30.3.16;rport=5060;branch=z9hG4bK8094.0045eaecd56ccafdd8b1a7abce0d2a82.0
Via: SIP/2.0/UDP 194.30.6.32;rport=5060;branch=z9hG4bK8094.92d538a79a25847bd20cd1d3fa78bcac.0
Via: SIP/2.0/UDP 212.106.213.37:5060;rport=5060;branch=z9hG4bK1855k330a8fghqsm53q1cd0000010.1
Call-ID: 1mmebggb3g01i0eafbki1iax442xd1fg@SoftX3000
From: <sip:639817681@212.106.213.37;user=phone>;tag=b0ikik0k-CC-48
To: <sip:912900324@194.30.6.32:5060;user=phone;user=phone>;tag=as616ecd38
CSeq: 2 BYE
Reason: Q.850;cause=16;text="normal call clearing"
Max-Forwards: 66
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 194.30.6.32:5060 (no NAT)
<--- Transmitting (no NAT) to 194.30.6.32:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 194.30.6.32;branch=z9hG4bK8094.16fe953123d66f6213332f91515e3e8f.0;received=194.30.6.32
Via: SIP/2.0/UDP 194.30.3.16;rport=5060;branch=z9hG4bK8094.0045eaecd56ccafdd8b1a7abce0d2a82.0
Via: SIP/2.0/UDP 194.30.6.32;rport=5060;branch=z9hG4bK8094.92d538a79a25847bd20cd1d3fa78bcac.0
Via: SIP/2.0/UDP 212.106.213.37:5060;rport=5060;branch=z9hG4bK1855k330a8fghqsm53q1cd0000010.1
From: <sip:639817681@212.106.213.37;user=phone>;tag=b0ikik0k-CC-48
To: <sip:912900324@194.30.6.32:5060;user=phone;user=phone>;tag=as616ecd38
Call-ID: 1mmebggb3g01i0eafbki1iax442xd1fg@SoftX3000
CSeq: 2 BYE
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Y la llamada termina. Según el proveedor SIP, el problema esta en que no le envío el BYE
Si la llamada entra directamente en una extensión (sin pasar por IVR) si se envía el BYE y la llamada termina normalmente.
Por cierto, esto no pasa en Issabel4 con Asterisk 16.
