Pourtant si je décroche et que l'appelant raccroche en premier ça fonctionne.
Mais dans un dialplan de ce style si l'on raccroche dès le départ pendant l'annonce Welcome1 ça sonne non stop et ça fini en timeout.
Code:
[from-pstn]
language=fr
exten => s,1, Answer
same => n, Wait(1)
same => n, Playback(welcome1)
same => n, Dial(SIP/gtab&SIP/iphone&Dahdi/3,25)
;same => n, Playback(vm-goodbye)
;same => n, Dial(SIP/gtab&SIP/iphone,5)
same => n,Goto(menu01,s,1)
same => n, Hangup()
[menu01]
exten => s,1, Background(message1)
exten => s,n, WaitExten(4)
exten => 1,1,Voicemail(222); 1
exten => 2,1,Voicemail(233); 2
exten => t,1,Playback(conf-errormenu)
exten => t,n,Playback(tt-somethingwrong)
exten => t,n,Playback(vm-goodbye)
exten => t,n,HangUp()
Cas ou l'appelant raccroche pendant le welcome1
En cli ça donne si l'on décroche:
Code:
serveur*CLI>
-- Starting simple switch on 'DAHDI/1-1'
[2012-01-15 15:22:24] WARNING[31780]: chan_dahdi.c:8939 ss_thread: CallerID returned with error on channel 'DAHDI/1-1'
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "1") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "welcome1") in new stack
-- <DAHDI/1-1> Playing 'welcome1.slin' (language 'fr')
-- Executing [s@from-pstn:5] Dial("DAHDI/1-1", "SIP/gtab&SIP/iphone&Dahdi/3,25") in new stack
== Using SIP RTP CoS mark 5
[2012-01-15 15:22:30] WARNING[31780]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[2012-01-15 15:22:30] WARNING[31780]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- Called 3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered DAHDI/1-1
-- Native bridging DAHDI/1-1 and DAHDI/3-1
-- Hungup 'DAHDI/3-1'
== Spawn extension (from-pstn, s, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
serveur*CLI>
Cas ou l'appelant raccroche pendant le welcome1
En cli ça donne si l'on ne fait rien:
Code:
-- Starting simple switch on 'DAHDI/1-1'
[2012-01-15 15:22:24] WARNING[31780]: chan_dahdi.c:8939 ss_thread: CallerID returned with error on channel 'DAHDI/1-1'
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "1") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "welcome1") in new stack
-- <DAHDI/1-1> Playing 'welcome1.slin' (language 'fr')
-- Executing [s@from-pstn:5] Dial("DAHDI/1-1", "SIP/gtab&SIP/iphone&Dahdi/3,25") in new stack
== Using SIP RTP CoS mark 5
[2012-01-15 15:22:30] WARNING[31780]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[2012-01-15 15:22:30] WARNING[31780]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- Called 3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered DAHDI/1-1
-- Native bridging DAHDI/1-1 and DAHDI/3-1
-- Hungup 'DAHDI/3-1'
== Spawn extension (from-pstn, s, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "1") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "welcome1") in new stack
-- <DAHDI/1-1> Playing 'welcome1.slin' (language 'fr')
-- Executing [s@from-pstn:5] Dial("DAHDI/1-1", "SIP/gtab&SIP/iphone&Dahdi/3,25") in new stack
== Using SIP RTP CoS mark 5
[2012-01-15 15:25:57] WARNING[476]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[2012-01-15 15:25:57] WARNING[476]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- Called 3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- Nobody picked up in 25000 ms
-- Hungup 'DAHDI/3-1'
-- Executing [s@from-pstn:6] Goto("DAHDI/1-1", "menu01,s,1") in new stack
-- Goto (menu01,s,1)
-- Executing [s@menu01:1] BackGround("DAHDI/1-1", "message1") in new stack
-- <DAHDI/1-1> Playing 'message1.slin' (language 'fr')
-- Executing [s@menu01:2] WaitExten("DAHDI/1-1", "4") in new stack
-- Timeout on DAHDI/1-1, going to 't'
-- Executing [t@menu01:1] Playback("DAHDI/1-1", "conf-errormenu") in new stack
-- <DAHDI/1-1> Playing 'conf-errormenu.slin' (language 'fr')
-- Executing [t@menu01:2] Playback("DAHDI/1-1", "tt-somethingwrong") in new stack
-- <DAHDI/1-1> Playing 'tt-somethingwrong.slin' (language 'fr')
-- Executing [t@menu01:3] Playback("DAHDI/1-1", "vm-goodbye") in new stack
-- <DAHDI/1-1> Playing 'vm-goodbye.slin' (language 'fr')
-- Executing [t@menu01:4] Hangup("DAHDI/1-1", "") in new stack
== Spawn extension (menu01, t, 4) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
Cas ou l'appelant raccroche en premier pendant une communication:
Code:
-- Hungup 'DAHDI/1-1'
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "1") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "welcome1") in new stack
-- <DAHDI/1-1> Playing 'welcome1.slin' (language 'fr')
-- Executing [s@from-pstn:5] Dial("DAHDI/1-1", "SIP/gtab&SIP/iphone&Dahdi/3,25") in new stack
== Using SIP RTP CoS mark 5
[2012-01-15 15:31:15] WARNING[1946]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[2012-01-15 15:31:15] WARNING[1946]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- Called 3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered DAHDI/1-1
-- Native bridging DAHDI/1-1 and DAHDI/3-1
-- Hungup 'DAHDI/3-1'
== Spawn extension (from-pstn, s, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
Cas ou l'appelé raccroche en premier:
Code:
-- Hungup 'DAHDI/1-1'
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "1") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "welcome1") in new stack
-- <DAHDI/1-1> Playing 'welcome1.slin' (language 'fr')
-- Executing [s@from-pstn:5] Dial("DAHDI/1-1", "SIP/gtab&SIP/iphone&Dahdi/3,25") in new stack
== Using SIP RTP CoS mark 5
[2012-01-15 15:33:00] WARNING[2134]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[2012-01-15 15:33:00] WARNING[2134]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-- Called 3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered DAHDI/1-1
-- Native bridging DAHDI/1-1 and DAHDI/3-1
-- Hungup 'DAHDI/3-1'
== Spawn extension (from-pstn, s, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
Je ne voi pas la différence entre les 2 derniers cas alors qu'il devrait y en avoir une non?