Hallo,
bei uns wurde gestern von ISDN auf SIP Trunk von der Telekom umgestellt.
Jetzt haben wir das Problem das alle Anrufe auf die Zentrale abgeworfen werden.
Was mich wundert ist das die Durchwahl nirgends im Log auftaucht, eigentlich habe ich die Durchwahl 18 angerufen.
[Jul 28 08:34:02] VERBOSE[5930][C-00000007] netsock2.c: == Using SIP RTP TOS bits 184
[Jul 28 08:34:02] VERBOSE[5930][C-00000007] netsock2.c: == Using SIP RTP CoS mark 5
[Jul 28 08:34:02] VERBOSE[5930][C-00000007] res_rtp_asterisk.c: > 0x7fdc8c02ecf0 -- Strict RTP learning after remote address set to: 217.0.132.148:10068
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@Telekom-incoming:1] Set("SIP/+4972xx70xxx-0000000c", "channelname=Telekom-incoming") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@Telekom-incoming:2] Set("SIP/+4972xx70xxx-0000000c", "lineconfigid=1376") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@Telekom-incoming:3] Goto("SIP/+4972xx70xxx-0000000c", "Telekom-incoming-manuell,70xxx,1") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Goto (Telekom-incoming-manuell,70xxx,1)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@Telekom-incoming-manuell:1] Goto("SIP/+4972xx70xxx-0000000c", "incoming,70xxx,1") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Goto (incoming,70xxx,1)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@incoming:1] Goto("SIP/+4972xx70xxx-0000000c", "calling,70xxx,1") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Goto (calling,70xxx,1)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@calling:1] GotoIf("SIP/+4972xx70xxx-0000000c", "0?4") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] pbx.c: -- Executing [70xxx@calling:2] AGI("SIP/+4972xx70xxx-0000000c", "agi:async,,1376,,,,,,,,,,,,,,") in new stack
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=70xxx : tester)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (X-UCI_CALLID)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (SIPAddHeader) Options: (X-UCI_CALLID:a8e07465-b36f-4841-98ba-5fe122245827)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=70xxx : tester)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(number)=00049xxxxx)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] res_agi.c: -- AGI Script Executing Application: (Dial) Options: (SIP/1016.ylnkt46,15,wWtT)
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] netsock2.c: == Using SIP RTP TOS bits 184
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] netsock2.c: == Using SIP RTP CoS mark 5
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] app_dial.c: -- Called SIP/1016.ylnkt46
[Jul 28 08:34:02] VERBOSE[16608][C-00000007] app_dial.c: -- SIP/1016.ylnkt46-0000000d is ringing
[Jul 28 08:34:07] VERBOSE[16608][C-00000007] pbx.c: == Spawn extension (calling, 70xxx, 2) exited non-zero on 'SIP/+4972xx70xxx-0000000c'
[Jul 28 08:34:07] VERBOSE[16608][C-00000007] pbx.c: -- Executing [h@calling:1] NoOp("SIP/+4972xx70xxx-0000000c", "HC 0") in new stack
[Jul 28 08:34:07] VERBOSE[16608][C-00000007] pbx.c: -- Executing [h@calling:2] Goto("SIP/+4972xx70xxx-0000000c", "_exit_,0") in new stack
[Jul 28 08:34:07] VERBOSE[16608][C-00000007] pbx.c: -- Goto (calling,_exit_,0)
manuelle Konfig wurde nicht verändert:
[Telekom-incoming]
exten => _.,1,Set(channelname=Telekom-incoming)
exten => _.,2,Set(lineconfigid=1376)
exten => _.,3,Goto(Telekom-incoming-manuell,${EXTEN},1)
[Telekom-incoming-manuell]
exten => _X.,1,Goto(incoming,${EXTEN},1)
exten => _+X.,1,Goto(incoming,${EXTEN},1)
Build-Version: 6.5.0.30
Hat jemand eine Idee?
Viele Grüße
Kay