hi:
i'm a new of asterisk voip server, i compiling without problem asterisk 1.4.18,
and other software and component.
i create two extension 20000 and 20100... and 30000 voicemailMain
but i can't call any extension this is the logs
/var/logs/asterisk/messages
[Apr 7 13:25:19] WARNING[24402] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
[Apr 7 13:26:27] WARNING[24407] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
[Apr 7 13:26:51] NOTICE[24408] cdr.c: CDR simple logging enabled.
[Apr 7 13:26:51] NOTICE[24408] loader.c: 144 modules will be loaded.
[Apr 7 13:26:51] WARNING[24408] res_smdi.c: No SMDI interfaces are available
to listen on, not starting SMDI listener.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: Starting AEL load process.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: calculated config
file name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: parsed config file
name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: checked config
file name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: compiled config
file name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: merged config file
name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] NOTICE[24408] pbx_ael.c: AEL load process: verified config
file name '/etc/asterisk/extensions.ael'.
[Apr 7 13:26:51] WARNING[24408] chan_iax2.c: Unable to open IAX timing
interface: No such file or directory
[Apr 7 13:27:02] WARNING[24439] file.c: File vm-login does not exist in any
format
[Apr 7 13:27:02] WARNING[24439] file.c: Unable to open vm-login (format 0x2
(gsm)): No such file or directory
[Apr 7 13:27:02] WARNING[24439] app_voicemail.c: Couldn't stream login file
[Apr 7 13:27:48] NOTICE[24419] chan_sip.c: Call from '20000' to extension
'500' rejected because extension not found.
[Apr 7 13:27:59] WARNING[24440] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
extensions.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
MACHINE1=SIP/20000
MACHINE2=SIP/20100
;My Extensions
[ejemplo]
;Yanier
exten=>20000,1,Dial(${MACHINE1},30,Tm)
exten=>20000,2,Hangup
exten=>20000,102,Voicemail(20000)
exten=>20000,103,Hangup
;Pedro
exten=>20100,1,Dial(${MACHINE2},30,Tm)
exten=>20100,2,Hangup
exten=>20100,102,Voicemail(20100)
exten=>20100,103,Hangup
;Other
exten=>30000,1,VoicemailMain
SIP.conf
;Test conf
[20000]
type=friend
secret=a20000b
qualify=yes
nat=no
canreinvite=no
context=ejemplo
[EMAIL PROTECTED]
callerid=Yanier
disallow=all
allow=gsm
[20100]
type=friend
secret=a20100b
qualify=yes
nat=no
canreinvite=no
context=ejemplo
[EMAIL PROTECTED]
callerid=Pedro
disallow=all
allow=gsm
voicemail.conf
[primerbuzon]
20000=>1234,Yanier,[EMAIL PROTECTED]
20100=>4321,Juan,[EMAIL PROTECTED]
someone can helpme
PD: Sorry for my bad english.
PD2: someone can explain how to install correct asterisk with some
configuration examples(only for pc lan).
Obe Provincial Ciego de Avila
Ave de los Deportes, esq. Circunvalación Norte
Telef: 200708
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