Asterisk SIP Out JA In Nein Dringend Hilfer benötigt auch gegen Geld

[Wageck]

ich hab hier einen Asterisk mit VICIDIAL eingerichtet und hänge jetzt an dem proplem das ich raus telefonieren kann nicht aber rein telefonieren kann.

komischer weise zeigt mir ein sip show registry nichts an
vor dem asterisk sitzt ein ipcop der die ports
udp 10000 - 20000 und TCP/UDP 5060 an den asterisk weiterleitet

da ich nicht weis was mein fehler ist hänge ich jetzt einfach mal meine sip.conf und meine extensions.conf an und hoffe ihr könnt mir nen tipp geben denn ich such jetzt schon 3 tage im forum und in google und alles was ich finde und probiere ändert nichts an meinem problem komischer weise erscheint auf der cli keinerlei meldung wenn ich den asterisk anrufe

sip.conf

[general]
context=default			; Default context for incoming calls
realm=dialer
bindport=5060	
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes
tos=184			; Set IP QoS to either a keyword or numeric val
tos=lowdelay			; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600			; Max length of incoming registration we allow
defaultexpiry=120		; Default length of incoming/outgoing registration
disallow=all	
allow=ulaw	
allow=alaw
allow=gsm
allow=ilbc	
language=de
dtmfmode = rfc2833

  
registertimeout=20		; retry registration calls every 20 seconds (default)

externip = 85.232.11.205	
localnet=192.168.100.0/255.255.255.0



;=================== SIP PROVIDER Registrierung ==========================================
;=================== ausgehende SIP Trunks ===============================================
; register SIP account on remote machine if using SIP trunks

;account for SIP trunking: Tel-Nr.: XXXX12799490
register => 10517128:[email protected]:5060/SIPtrunk490

[SIPtrunk490]
type=friend
username=10517128
fromuser=10517128
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXX12799491
register => 10517129:[email protected]:5060/SIPtrunk


[SIPtrunk]
type=friend
username=10517129
fromuser=10517129
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXX12799492
register => 10517130:[email protected]:5060/SIPtrunk492

[SIPtrunk492]
type=friend
username=10517130
fromuser=10517130
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXX12799493
register => 10517131:[email protected]:5060/SIPtrunk493

[SIPtrunk493]
type=friend
username=10517131
fromuser=10517131
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXX12799494
;register => 10517132:[email protected]:5060/SIPtrunk494

[SIPtrunk494]
type=friend
username=10517132
fromuser=10517132
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXXXXX12799495
register => 10517133:[email protected]:5060/SIPtrunk495

[SIPtrunk495]
type=friend
username=10517133
fromuser=10517133
secret=hz3hJ9R
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXXX12799496
register => 10517134:[email protected]:5060/SIPtrunk496


[SIPtrunk496]
type=friend
username=10517134
fromuser=10517134
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXXXX12799497
;register => 10517135:[email protected]:5060/SIPtrunk497

[SIPtrunk497]
type=friend
username=10517135
fromuser=10517135
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXXXX12799498
register => 10517136:[email protected]:5060/SIPtrunk498

[SIPtrunk498]
type=friend
username=10517136
fromuser=10517136
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

;account for SIP trunking: Tel-Nr.: XXXXX12799499
register => 10517137:[email protected]:5060/SIPtrunk499
;
;setup account for SIP trunking:
[SIPtrunk499]
type=friend
username=10517137
fromuser=10517137
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
context=inbound
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc



[inbound] 
type=peer
fromdomain=sip.24x7-business.de
host=sip.24x7-business.de
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=inbound
;===================SIP Client Registrierung ==========================================
[1]
callerid=Agent 1 <1>
host=dynamic
domain=hepex.local
user=1
secret=geheim
type=friend
mailbox=10
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[10]
callerid=Agent 10 <10>
host=dynamic
domain=hepex.local
user=10
secret=geheim
type=friend
mailbox=10
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[11]
callerid=Agent 11 <11>
host=dynamic
domain=hepex.local
user=11
secret=geheim
type=friend
mailbox=11
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[12]
callerid=Agent 12 <12>
host=dynamic
domain=hepex.local
user=12
secret=geheim
type=friend
mailbox=12
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[13]
callerid=Agent 13 <13>
host=dynamic
domain=hepex.local
user=13
secret=geheim
type=friend
mailbox=13
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[14]
callerid=Agent 14 <14>
host=dynamic
domain=hepex.local
user=14
secret=geheim
type=friend
mailbox=14
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[15]
callerid=Agent 15 <15>
host=dynamic
domain=hepex.local
user=15
secret=geheim
type=friend
mailbox=15
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[16]
callerid=Agent 16 <16>
host=dynamic
domain=hepex.local
user=16
secret=geheim
type=friend
mailbox=16
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[17]
callerid=Agent 17 <17>
host=dynamic
domain=hepex.local
user=17
secret=geheim
type=friend
mailbox=17
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[18]
callerid=Agent 18 <18>
host=dynamic
domain=hepex.local
user=18
secret=geheim
type=friend
mailbox=18
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[19]
callerid=Agent 19 <19>
host=dynamic
domain=hepex.local
user=19
secret=geheim
type=friend
mailbox=19
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[20]
callerid=Agent 20 <20>
host=dynamic
domain=hepex.local
user=20
secret=geheim
type=friend
mailbox=20
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[21]
callerid=Agent 21 <21>
host=dynamic
domain=hepex.local
user=21
secret=geheim
type=friend
mailbox=21
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[22]
callerid=Agent 22 <22>
host=dynamic
domain=hepex.local
user=22
secret=geheim
type=friend
mailbox=22
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[23]
callerid=Agent 23 <23>
host=dynamic
domain=hepex.local
user=23
secret=geheim
type=friend
mailbox=23
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[24]
callerid=Agent 24 <24>
host=dynamic
domain=hepex.local
user=24
secret=geheim
type=friend
mailbox=24
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[25]
callerid=Agent 25 <25>
host=dynamic
domain=hepex.local
user=25
secret=geheim
type=friend
mailbox=25
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[26]
callerid=Agent 26 <26>
host=dynamic
domain=hepex.local
user=26
secret=geheim
type=friend
mailbox=26
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[27]
callerid=Agent 27 <27>
host=dynamic
domain=hepex.local
user=27
secret=geheim
type=friend
mailbox=27
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[28]
callerid=Agent 28 <28>
host=dynamic
domain=hepex.local
user=28
secret=geheim
type=friend
mailbox=28
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[29]
callerid=Agent 29 <29>
host=dynamic
domain=hepex.local
user=29
secret=geheim
type=friend
mailbox=29
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

[30]
callerid=Agent 30 <30>
host=dynamic
domain=hepex.local
user=30
secret=geheim
type=friend
mailbox=30
nat=yes
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp				; Console interface for demo








[inbound]
exten => SIPtrunk490,1,Ringing
exten => SIPtrunk490,2,Answer
exten => SIPtrunk490,3,Dial,SIP/17&SIP/1|30|r

exten => 10517128,1,Ringing
exten => 10517128,2,Answer
exten => 10517128,3,Dial,SIP/17&SIP/1|30|r

exten => s,1,Ringing
exten => s,2,Answer
exten => s,3,Dial,SIP/17&SIP/1|30|r



[default]
;======== Eigener Wählplan ===========
; Mailbox Nr 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)

;Rauswählen über SIP
exten => _XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXX.,2,Dial(SIP/${EXTEN}@SIPtrunk,55,tTo)
exten => _XXX.,3,Hangup

;Intern Telefonieren
exten => _XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XX,2,Dial,SIP/${EXTEN}|55|Ttr
exten => _XX,3,Hangup

;##### This 'h' exten is VERY important for VICIDIAL usage, 
;##### you will have problems if it is not in your dialplan!
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten => h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}))


; parameters for agi-VDADcloser.agi (2 fields separated by five dashes "-----"):
; 1. the full extension formatted by VICIDIAL for internal transfers * separated
; 2. the word START to denote the beginning of the acceptance of the transfer
; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover or IAX channel]
exten => _90009.,1,Answer                  ; Answer the line
exten => _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-----START)
exten => _90009.,3,Hangup

;######------ START meetme.conf additions for conferences ------######
conf => 8600001
conf => 8600002
conf => 8600003
conf => 8600004
conf => 8600005
conf => 8600006
conf => 8600007
conf => 8600008
conf => 8600009
conf => 8600010
conf => 8600011
conf => 8600012
conf => 8600013
conf => 8600014
conf => 8600015
conf => 8600016
conf => 8600017
conf => 8600018
conf => 8600019
conf => 8600020
conf => 8600021
conf => 8600022
conf => 8600023
conf => 8600024
conf => 8600025
conf => 8600026
conf => 8600027
conf => 8600028
conf => 8600029
conf => 8600051
conf => 8600052
conf => 8600053
conf => 8600054
conf => 8600055
conf => 8600056
conf => 8600057
conf => 8600058
conf => 8600059
conf => 8600060
conf => 8600061
conf => 8600062
conf => 8600063
conf => 8600064
conf => 8600065
conf => 8600066
conf => 8600067
conf => 8600068
conf => 8600069
conf => 8600070
conf => 8600071
conf => 8600072
conf => 8600073
conf => 8600074
conf => 8600075
conf => 8600076
conf => 8600077
conf => 8600078
conf => 8600079
conf => 8600080
conf => 8600081
conf => 8600082
conf => 8600083
conf => 8600084
conf => 8600085
conf => 8600086
conf => 8600087
conf => 8600088
conf => 8600089
conf => 8600090
conf => 8600091
conf => 8600092
conf => 8600093
conf => 8600094
conf => 8600095
conf => 8600096
conf => 8600097
conf => 8600098
conf => 8600099
conf => 8600100
conf => 8600101
conf => 8600102
conf => 8600103
conf => 8600104
conf => 8600105
conf => 8600106
conf => 8600107
conf => 8600108
conf => 8600109
conf => 8600110
conf => 8600111
conf => 8600112
conf => 8600113
conf => 8600114
conf => 8600115
conf => 8600116
conf => 8600117
conf => 8600118
conf => 8600119
conf => 8600120
conf => 8600121
conf => 8600122
conf => 8600123
conf => 8600124
conf => 8600125
conf => 8600126
conf => 8600127
conf => 8600128
conf => 8600129
conf => 8600130
conf => 8600131
conf => 8600132
conf => 8600133
conf => 8600134
conf => 8600135
conf => 8600136
conf => 8600137
conf => 8600138
conf => 8600139
conf => 8600140
conf => 8600141
conf => 8600142
conf => 8600143
conf => 8600144
conf => 8600145
conf => 8600146
conf => 8600147
conf => 8600148
conf => 8600149
conf => 8600150
conf => 8600151
conf => 8600152
conf => 8600153
conf => 8600154
conf => 8600155
conf => 8600156
conf => 8600157
conf => 8600158
conf => 8600159
conf => 8600160
conf => 8600161
conf => 8600162
conf => 8600163
conf => 8600164
conf => 8600165
conf => 8600166
conf => 8600167
conf => 8600168
conf => 8600169
conf => 8600170
conf => 8600171
conf => 8600172
conf => 8600173
conf => 8600174
conf => 8600175
conf => 8600176
conf => 8600177
conf => 8600178
conf => 8600179
conf => 8600180
conf => 8600181
conf => 8600182
conf => 8600183
conf => 8600184
conf => 8600185
conf => 8600186
conf => 8600187
conf => 8600188
conf => 8600189
conf => 8600190
conf => 8600191
conf => 8600192
conf => 8600193
conf => 8600194
conf => 8600195
conf => 8600196
conf => 8600197
conf => 8600198
conf => 8600199
conf => 8600200
;######------ END meetme.conf additions for conferences ------######

;######------ START extensions.conf additions for agc conferences ------######
exten => 8600001,1,Conference(8600001|q)
exten => 8600002,1,Conference(8600002|q)
exten => 8600003,1,Conference(8600003|q)
exten => 8600004,1,Conference(8600004|q)
exten => 8600005,1,Conference(8600005|q)
exten => 8600006,1,Conference(8600006|q)
exten => 8600007,1,Conference(8600007|q)
exten => 8600008,1,Conference(8600008|q)
exten => 8600009,1,Conference(8600009|q)
exten => 8600010,1,Conference(8600010|q)
exten => 8600011,1,Conference(8600011|q)
exten => 8600012,1,Conference(8600012|q)
exten => 8600013,1,Conference(8600013|q)
exten => 8600014,1,Conference(8600014|q)
exten => 8600015,1,Conference(8600015|q)
exten => 8600016,1,Conference(8600016|q)
exten => 8600017,1,Conference(8600017|q)
exten => 8600018,1,Conference(8600018|q)
exten => 8600019,1,Conference(8600019|q)
exten => 8600020,1,Conference(8600020|q)
exten => 8600021,1,Conference(8600021|q)
exten => 8600022,1,Conference(8600022|q)
exten => 8600023,1,Conference(8600023|q)
exten => 8600024,1,Conference(8600024|q)
exten => 8600025,1,Conference(8600025|q)
exten => 8600026,1,Conference(8600026|q)
exten => 8600027,1,Conference(8600027|q)
exten => 8600028,1,Conference(8600028|q)
exten => 8600029,1,Conference(8600029|q)
;######------ END extensions.conf additions for agc conferences ------######

;######------ START extensions.conf changes for VD conf ------######
exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten => _X38600XXX,2,Hangup

exten => 8300,1,Hangup

exten => 8600051,1,Conference(8600051)
exten => 8600052,1,Conference(8600052)
exten => 8600053,1,Conference(8600053)
exten => 8600054,1,Conference(8600054)
exten => 8600055,1,Conference(8600055)
exten => 8600056,1,Conference(8600056)
exten => 8600057,1,Conference(8600057)
exten => 8600058,1,Conference(8600058)
exten => 8600059,1,Conference(8600059)
exten => 8600060,1,Conference(8600060)
exten => 8600061,1,Conference(8600061)
exten => 8600062,1,Conference(8600062)
exten => 8600063,1,Conference(8600063)
exten => 8600064,1,Conference(8600064)
exten => 8600065,1,Conference(8600065)
exten => 8600066,1,Conference(8600066)
exten => 8600067,1,Conference(8600067)
exten => 8600068,1,Conference(8600068)
exten => 8600069,1,Conference(8600069)
exten => 8600070,1,Conference(8600070)
exten => 8600071,1,Conference(8600071)
exten => 8600072,1,Conference(8600072)
exten => 8600073,1,Conference(8600073)
exten => 8600074,1,Conference(8600074)
exten => 8600075,1,Conference(8600075)
exten => 8600076,1,Conference(8600076)
exten => 8600077,1,Conference(8600077)
exten => 8600078,1,Conference(8600078)
exten => 8600079,1,Conference(8600079)
exten => 8600080,1,Conference(8600080)
exten => 8600081,1,Conference(8600081)
exten => 8600082,1,Conference(8600082)
exten => 8600083,1,Conference(8600083)
exten => 8600084,1,Conference(8600084)
exten => 8600085,1,Conference(8600085)
exten => 8600086,1,Conference(8600086)
exten => 8600087,1,Conference(8600087)
exten => 8600088,1,Conference(8600088)
exten => 8600089,1,Conference(8600089)
exten => 8600090,1,Conference(8600090)
exten => 8600091,1,Conference(8600091)
exten => 8600092,1,Conference(8600092)
exten => 8600093,1,Conference(8600093)
exten => 8600094,1,Conference(8600094)
exten => 8600095,1,Conference(8600095)
exten => 8600096,1,Conference(8600096)
exten => 8600097,1,Conference(8600097)
exten => 8600098,1,Conference(8600098)
exten => 8600099,1,Conference(8600099)
exten => 8600100,1,Conference(8600100)
exten => 8600101,1,Conference(8600101)
exten => 8600102,1,Conference(8600102)
exten => 8600103,1,Conference(8600103)
exten => 8600104,1,Conference(8600104)
exten => 8600105,1,Conference(8600105)
exten => 8600106,1,Conference(8600106)
exten => 8600107,1,Conference(8600107)
exten => 8600108,1,Conference(8600108)
exten => 8600109,1,Conference(8600109)
exten => 8600110,1,Conference(8600110)
exten => 8600111,1,Conference(8600111)
exten => 8600112,1,Conference(8600112)
exten => 8600113,1,Conference(8600113)
exten => 8600114,1,Conference(8600114)
exten => 8600115,1,Conference(8600115)
exten => 8600116,1,Conference(8600116)
exten => 8600117,1,Conference(8600117)
exten => 8600118,1,Conference(8600118)
exten => 8600119,1,Conference(8600119)
exten => 8600120,1,Conference(8600120)
exten => 8600121,1,Conference(8600121)
exten => 8600122,1,Conference(8600122)
exten => 8600123,1,Conference(8600123)
exten => 8600124,1,Conference(8600124)
exten => 8600125,1,Conference(8600125)
exten => 8600126,1,Conference(8600126)
exten => 8600127,1,Conference(8600127)
exten => 8600128,1,Conference(8600128)
exten => 8600129,1,Conference(8600129)
exten => 8600130,1,Conference(8600130)
exten => 8600131,1,Conference(8600131)
exten => 8600132,1,Conference(8600132)
exten => 8600133,1,Conference(8600133)
exten => 8600134,1,Conference(8600134)
exten => 8600135,1,Conference(8600135)
exten => 8600136,1,Conference(8600136)
exten => 8600137,1,Conference(8600137)
exten => 8600138,1,Conference(8600138)
exten => 8600139,1,Conference(8600139)
exten => 8600140,1,Conference(8600140)
exten => 8600141,1,Conference(8600141)
exten => 8600142,1,Conference(8600142)
exten => 8600143,1,Conference(8600143)
exten => 8600144,1,Conference(8600144)
exten => 8600145,1,Conference(8600145)
exten => 8600146,1,Conference(8600146)
exten => 8600147,1,Conference(8600147)
exten => 8600148,1,Conference(8600148)
exten => 8600149,1,Conference(8600149)
exten => 8600150,1,Conference(8600150)
exten => 8600151,1,Conference(8600151)
exten => 8600152,1,Conference(8600152)
exten => 8600153,1,Conference(8600153)
exten => 8600154,1,Conference(8600154)
exten => 8600155,1,Conference(8600155)
exten => 8600156,1,Conference(8600156)
exten => 8600157,1,Conference(8600157)
exten => 8600158,1,Conference(8600158)
exten => 8600159,1,Conference(8600159)
exten => 8600160,1,Conference(8600160)
exten => 8600161,1,Conference(8600161)
exten => 8600162,1,Conference(8600162)
exten => 8600163,1,Conference(8600163)
exten => 8600164,1,Conference(8600164)
exten => 8600165,1,Conference(8600165)
exten => 8600166,1,Conference(8600166)
exten => 8600167,1,Conference(8600167)
exten => 8600168,1,Conference(8600168)
exten => 8600169,1,Conference(8600169)
exten => 8600170,1,Conference(8600170)
exten => 8600171,1,Conference(8600171)
exten => 8600172,1,Conference(8600172)
exten => 8600173,1,Conference(8600173)
exten => 8600174,1,Conference(8600174)
exten => 8600175,1,Conference(8600175)
exten => 8600176,1,Conference(8600176)
exten => 8600177,1,Conference(8600177)
exten => 8600178,1,Conference(8600178)
exten => 8600179,1,Conference(8600179)
exten => 8600180,1,Conference(8600180)
exten => 8600181,1,Conference(8600181)
exten => 8600182,1,Conference(8600182)
exten => 8600183,1,Conference(8600183)
exten => 8600184,1,Conference(8600184)
exten => 8600185,1,Conference(8600185)
exten => 8600186,1,Conference(8600186)
exten => 8600187,1,Conference(8600187)
exten => 8600188,1,Conference(8600188)
exten => 8600189,1,Conference(8600189)
exten => 8600190,1,Conference(8600190)
exten => 8600191,1,Conference(8600191)
exten => 8600192,1,Conference(8600192)
exten => 8600193,1,Conference(8600193)
exten => 8600194,1,Conference(8600194)
exten => 8600195,1,Conference(8600195)
exten => 8600196,1,Conference(8600196)
exten => 8600197,1,Conference(8600197)
exten => 8600198,1,Conference(8600198)
exten => 8600199,1,Conference(8600199)
exten => 8600200,1,Conference(8600200)
; quiet entry and leaving conferences for VICIDIAL
exten => _78600XXX,1,Conference(${EXTEN:1}|q)
; quiet monitor extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Conference(${EXTEN:1}|mq)

;######------ END extensions.conf changes for VD conf ------######

;######------ START extensions.conf other additions ------######
; park channel for client GUI parking, hangup after 30 minutes
;    create a GSM formatted audio file named "park.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup 
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup 

; park channel for client GUI conferencing, hangup after 30 minutes
;    create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
;    create a GSM formatted audio file complies with safe harbor rules
;    and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERIDNAME})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
;    replace conf with the message file you want to leave
exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

; this is used to allow the GUI to send you directly into voicemail
;     don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail
;     don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(2)
exten => _85026666666666.,2,Voicemail(${EXTEN:14})
exten => _85026666666666.,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
;    sends the digits to be played in the callerID field
;    sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; prompt recording AGI script, ID is 4321
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi)
exten => 8168,3,Hangup

; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

;#### VDAD STANDARD TRANSFER ENTRIES ####
; VICIDIAL_auto_dialer transfer script for no-agent campaigns:
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDADtransferBROADCAST.agi,${EXTEN})
exten => 8364,4,AGI(agi-VDADtransferBROADCAST.agi,${EXTEN})
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,2,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,2,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,2,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,3,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,4,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,2,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,3,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,2,AMD(3500|1500|300|5000|120|50|5|256)
exten => 8369,3,AGI(VD_amd.agi,${EXTEN})
exten => 8369,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,6,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,2,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

 

[toldap]

übernehme doch mal testweise deine Config unter dem inbound Context für den
default Context.
Ich hatte bei Sip-Incoming schon einigemale das Problem, dass der "falsche" context verwendet wurde. Vorallem bei mehreren Sip-Trunks.
Dies ist auch ersichtlich, wenn Du sip debug einschaltest.

MfG

 

[Wageck]

das hab ich schon versucht gibt es eine möglichkeit irgend wie den fehler zu finden ?

 

[toldap]

Schalte doch mal den DEBUG Modus für Sip ein, versuch dich selber über die Sip-Leitung anzurufen und poste hier was der DEBUG bringt.

 

[Wageck]

ich hab jetzt die kiste neugsetartet
set verboe = 0 gemacht und dann sip debug
und noch befor ich ich angerufen habe ist folgende erschienen

Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK785d8975;rport
From: "asterisk" <sip:[email protected]>;tag=as6fdb2413
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Mar 2007 21:02:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK28745c8e;rport
From: "asterisk" <sip:[email protected]>;tag=as7c906516
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Mar 2007 21:02:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK7a6d6ac0;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as435a70b6
To: <sip:sip.24x7-business.de>;tag=as1b8c2439
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK40709636;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7969427b
To: <sip:sip.24x7-business.de>;tag=as3262177b
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK7b6ac39b;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as2e633c40
To: <sip:sip.24x7-business.de>;tag=as34c91d6e
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK09da6c16;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as6512ea07
To: <sip:sip.24x7-business.de>;tag=as72098f8e
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK3e88b7f4;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as2dd28e34
To: <sip:sip.24x7-business.de>;tag=as4cf1eae3
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK46ece981;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as74d2d2e8
To: <sip:sip.24x7-business.de>;tag=as7e88543b
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK03643d4d;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as41830b84
To: <sip:sip.24x7-business.de>;tag=as18c0f0fa
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK4822cc0d;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7af0c048
To: <sip:sip.24x7-business.de>;tag=as691e330c
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK785d8975;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as6fdb2413
To: <sip:sip.24x7-business.de>;tag=as4c13203b
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK28745c8e;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7c906516
To: <sip:sip.24x7-business.de>;tag=as527acf60
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>

im mom sind 2 sip softphones angemeldet einmal mit der nr 1 und einmal die nr 17

folgendes wurde angezeigt als ich versucht habe anzurufen

Reliably Transmitting (no NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK699c5e58;rport
From: "asterisk" <sip:[email protected]>;tag=as6d93524a
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Mar 2007 21:05:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK4f0bd0e8;rport
From: "asterisk" <sip:[email protected]>;tag=as5ee13827
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Mar 2007 21:05:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK23698213;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as63f2c375
To: <sip:sip.24x7-business.de>;tag=as4489a2bf
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK56c49fe6;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as47fb616f
To: <sip:sip.24x7-business.de>;tag=as530744da
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK3cbcf6f1;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as020ed4e8
To: <sip:sip.24x7-business.de>;tag=as786cf558
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK0a3703bf;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as140dce85
To: <sip:sip.24x7-business.de>;tag=as082e02c7
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK73f840f6;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7d39bd2b
To: <sip:sip.24x7-business.de>;tag=as3adcfac1
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK07014846;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7cea6d80
To: <sip:sip.24x7-business.de>;tag=as6484d474
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK6de7aea4;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as3d1d12d7
To: <sip:sip.24x7-business.de>;tag=as6669ba4f
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK1db58f19;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1c27435e
To: <sip:sip.24x7-business.de>;tag=as31af9ccc
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK699c5e58;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as6d93524a
To: <sip:sip.24x7-business.de>;tag=as47765556
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK4f0bd0e8;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as5ee13827
To: <sip:sip.24x7-business.de>;tag=as29df07a0
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>

allerdings scheint es mir als ob der anruf garnicht beim asterisk ankommt und mir ist SIP/2.0 404 Not Found aufgefallen aber was heist das ?

 

[toldap]

Verwende mal die Option
nat=yes
auch bei den Sip-Trunks. Oder hast Du gute Gründe, diesen Wert nicht gesetzt zu haben?

Was die Meldung "SIP/2.0 404 Not Found" genau zu beseuten hat, kann ich leider auch nicht sagen. Ich bekahm solche Logs, wennn a. die Sip.conf Fehlerhaft war oder b. der Dialplan.... Ich denke kaum, dass man diese Meldung einem bestimmten Fehler zuweisen kann.

Von einem fälschlichen Context der angewählt wird ist allerdings nichts zu sehen. Ich gehe mal davon aus, dass das Problem noch einige "Layers" tiefer ist. Versuch mal nat=yes ....

 

[Wageck]

ok ich hab jetzt überall nat=yes eingetragen aber es scheint sich nix zu ändern muss ich vieleicht weitere ports weiterleiten ?

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK24318996;rport
From: "asterisk" <sip:[email protected]>;tag=as10b4223c
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Mar 2007 21:37:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK530d19fa;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as338eec69
To: <sip:sip.24x7-business.de>;tag=as7a0b49c5
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK74466bca;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as4d2ac2db
To: <sip:sip.24x7-business.de>;tag=as3ad01b00
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK0f5d18e9;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as56062c3d
To: <sip:sip.24x7-business.de>;tag=as6b92b6e6
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK5678245e;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as2186184c
To: <sip:sip.24x7-business.de>;tag=as1b749018
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK05dac84d;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as104dbc64
To: <sip:sip.24x7-business.de>;tag=as429a0656
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK7eb07998;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as711d50ea
To: <sip:sip.24x7-business.de>;tag=as5ed9d0d1
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK12d74293;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1b2e496a
To: <sip:sip.24x7-business.de>;tag=as17fabefc
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'

<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK7e7a9c2e;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as30ff8a7f
To: <sip:sip.24x7-business.de>;tag=as7f979140
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK153b86d1;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as568ed39e
To: <sip:sip.24x7-business.de>;tag=as66bc7901
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK24318996;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as10b4223c
To: <sip:sip.24x7-business.de>;tag=as5380d3e8
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>
<-- SIP read from 192.168.100.253:8340:


--- (0 headers 0 lines) Nat keepalive ---
dialer*CLI>

 

[toldap]

Muss leider für einige Stunden weg, werde anschliessend mal meine Config und meine Routing Einträge (wie es bei mir funkt.) hier posten.

MfG

 

[Wageck]

ok danke das du hilfts ich bin morgen wieder on

 

[toldap]

So, wieder online...

Habe Deine Config etwas genauer angeschaut:
Ich gehe mal davon aus, dass du auf die Nr.XXXX12799490 anrufen möchtest.
Als erster fällt mir dein Registry auf:

register => 10517128:[email protected]:5060/SIPtrunk490
Ich denke dieser müsste in diser Form sein:
register => usernameassword@register-server/telephonnummer
Also für Dich etwa so:
register => 10517128:[email protected]/12799490

Zu dieser würde dann in deiner sip.conf ja der Context [inbound] gehören.
Ändere doch mal unter diesem Context, den context für die extensions.conf, also;

context=inbound => ändern in => context=inboundbusiness (nicht gleich wie der Context in der sip.conf. Bedenke zusätzlich, dass dieser Context (Also der in der sip.conf "inbound") für jegliche Deine inbound-accounts von diesem Provider geltet. Zudem muss diser Context am Schluss deiner sip.conf stehen, d.h. vorallem nach allen Contexten die sich auf diesen Provider beziehen.

Jetzt zu Deiner extensions.conf.
Mach doch einfach mal einen neuen Context namens inboundbusiness, darunter kommt ganz einfach mal:
exten => 12799490,1,Dial,SIP/17|30|r

Somit solltest Du dich über die Nummer 12799490 auf deinem Sip-Client anrufen können. Allerdings momentan nur auf dieser Nummer. Wenn Du auf der CLI jetzt einen
sip show registry machst, müsste dieser Accoungt registriert sein. Falls nicht, stimmt Dein register Aufruf in der sip.conf nicht.

Firewall:
Damit Du sicher bist, dass es nicht am Cop liegt, vieleicht mal "optimistisch"
die Ports 5060 - 5086 UDP und TCP durchrouten.

 

[Wageck]

ok ich han meine xtension.conf jetzt wie gesagt das bisherige inbound durch inboundbusiness geändert

[inboundbusiness]
exten => 004934112799490,1,Dial,SIP/17|30|r

und in der sip.conf fogendes geändert

;account for SIP trunking: Tel-Nr.: 004934112799490
register => 10517128:[email protected]/004934112799490
;
;setup account for SIP trunking:
[SIPtrunk490]
type=friend
username=10517128
fromuser=10517128
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=inbound



[inbound] 
type=peer
fromdomain=sip.24x7-business.de
host=sip.24x7-business.de
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=inboundbusiness

und die gesagten ports hab ich auch weitergeleitet

allerdings zeigt ein sip show registry immer noch nichts

 

[toldap]

Ändere mal den reigster wie folgt;

register => 10517128:[email protected]/004934112799490
in
register => 10517128:[email protected]/2799490

Bist Du sicher dass dein Benutzername & passwd, stimmen? ggf. mit einem softphone oder so schnell kontrollieren.

 

[toldap]

Nicht etwa noch eine software-FW am start, oder?

 

[Wageck]

Eine Software FW ist keine da

den register werd ich mal ändern und da ich depp gerade gesehen habe das ich das pw mit kopiert habe wäre es nett wenn du es in deinem beitrag ändern würdest

die zugangsdaten stimmen da ich sie auch für ausgehend nutze

 

[Wageck]

auch ein ändern bringt nix

 

[Wageck]

mach ich eigentlich was falsch das mir fast niemand hilft ?

 

[Wageck]

sorry doppel post da mir aber gerade was aufgefallen ist poste ich das direkt hier

ich kann per telnet ne ferbindung auf den webserver auf der maschiene machen also auf port 80 aber eine verbindung auf port 5060 geht nicht bedeutet das was ?

 

[toldap]

Ich würde weiter am register Code arbeiten, versuch doch mal:

register => 10517128:[email protected]/10517128

Zu der Verbindung auf Port 5060. habs grad selbst ausprobiert, krieg mit telnet auch
keine Verbindung zu stande, sip läuft jedoch einwandfrei.

 

[Wageck]

soll ich mal alle register auskommentieren bis auf einen und dann weiter testen ?

 

[Wageck]

ich hab jetzt in der sip conf nur noch einen registrar drin und die xtension conf auch etwas bearbeitet jetzt wird mir in einem sip debug folgendes gezeigt zusätzlich dazu zeigt mir sip show registry und sip show peers folgendes an

sip.conf registry

; register SIP account on remote machine if using SIP trunks
;account for SIP trunking: Tel-Nr.: 004934112799491
register => 10517129:[email protected]:5060/10517129

[10517129]
type=friend
username=10517129
fromuser=10517129
secret=geheim
host=sip.24x7-business.de
dtmfmode=inband
qualify=1000
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc


[inbound] 
type=peer
fromdomain=sip.24x7-business.de
host=sip.24x7-business.de
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=inboundbusiness

extensions.conf

[inboundbusiness]
exten => 10517129,1,Ringing
exten => 10517129,2,Answer
exten => 10517129,3,Dial,SIP/17&SIP/1|30|r

sip show registry

dialer*CLI> sip show registry
Host                            Username       Refresh State
dialer*CLI>

Sip show peers

Name/username              Host            Dyn Nat ACL Port     Status
30                         (Unspecified)    D   N      0        Unmonitored
29                         (Unspecified)    D   N      0        Unmonitored
28                         (Unspecified)    D   N      0        Unmonitored
27                         (Unspecified)    D   N      0        Unmonitored
26                         (Unspecified)    D   N      0        Unmonitored
25                         (Unspecified)    D   N      0        Unmonitored
24                         (Unspecified)    D   N      0        Unmonitored
23                         (Unspecified)    D   N      0        Unmonitored
22                         (Unspecified)    D   N      0        Unmonitored
21                         (Unspecified)    D   N      0        Unmonitored
20                         (Unspecified)    D   N      0        Unmonitored
19                         (Unspecified)    D   N      0        Unmonitored
18                         (Unspecified)    D   N      0        Unmonitored
17/17                      10.23.14.6       D   N      5070     Unmonitored
16                         (Unspecified)    D   N      0        Unmonitored
15/15                      192.168.100.195  D   N      7818     Unmonitored
14                         (Unspecified)    D   N      0        Unmonitored
13                         (Unspecified)    D   N      0        Unmonitored
12                         (Unspecified)    D   N      0        Unmonitored
11                         (Unspecified)    D   N      0        Unmonitored
10/10                      192.168.100.197  D   N      7836     Unmonitored
1/1                        192.168.100.253  D   N      8340     Unmonitored
inbound                    217.24.217.52        N      5060     Unmonitored
10517129/10517129          217.24.217.52        N      5060     OK (21 ms)
24 sip peers [24 online , 0 offline]

Sip Debug

12 headers, 0 lines
Reliably Transmitting (NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK30e0648d;rport
From: "asterisk" <sip:[email protected]>;tag=as41f0de98
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 13 Mar 2007 13:40:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK30e0648d;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as41f0de98
To: <sip:sip.24x7-business.de>;tag=as3c869259
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>

12 headers, 0 lines
Reliably Transmitting (NAT) to 217.24.217.52:5060:
OPTIONS sip:sip.24x7-business.de SIP/2.0
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK30e0648d;rport
From: "asterisk" <sip:[email protected]>;tag=as41f0de98
To: <sip:sip.24x7-business.de>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 13 Mar 2007 13:40:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
dialer*CLI>
<-- SIP read from 217.24.217.52:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.232.11.205:5060;branch=z9hG4bK30e0648d;received=85.232.11.205;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as41f0de98
To: <sip:sip.24x7-business.de>;tag=as3c869259
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: PBX-network SERVER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


--- (10 headers 0 lines) ---
Destroying call '[email protected]'
dialer*CLI>