asterisk users - Apr 2008 - new instalation os asterisk

On Mon, 7 Apr 2008, Yanier Salazar Sanchez wrote:
> [Apr 7 13:25:19] WARNING[24402] app_dial.c: Unable to create channel of 
> type 'SIP' (cause 3 - No route to destination)
Whatever device was the destination could not be found.
> [Apr 7 13:26:51] WARNING[24408] chan_iax2.c: Unable to open IAX timing 
> interface: No such file or directory
You do not have zaptel installed. Zaptel needs to be compiled before 
Asterisk.
> [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
You do not have the "asterisk sounds" installed.
> [Apr 7 13:27:48] NOTICE[24419] chan_sip.c: Call from '20000' to 
> extension '500' rejected because extension not found.
You do not have an extension 500 defined in the context ejemplo.

You may want to do a bit more reading before posting again and using up 
your "karma." You may want to install a "all-in-one" package
like Trixbox
before you attempt to build one from scratch.

Building from scratch will be a much more involved (and satisfying) 
experience, but starting with an all-in-one will get you something to play 
with faster.

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000

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
mailbox=20000 at ejemplobuzon
callerid=Yanier
disallow=all
allow=gsm

[20100]
type=friend
secret=a20100b
qualify=yes
nat=no
canreinvite=no
context=ejemplo
mailbox=20100 at ejemplobuzon
callerid=Pedro
disallow=all
allow=gsm

voicemail.conf
[primerbuzon]
20000=>1234,Yanier,yanier at micorreo.com
20100=>4321,Juan,juan at micorreo.com


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. Circunvalaci?n Norte
Telef: 200708
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