[asterisk-users] calls are not going thru e1 line
Albert
alberton at wp.pl
Mon Feb 21 07:52:58 CST 2011
Hi Andrew,
I am using current versions of software, find below versions:
1.) asterisk
voice:~# asterisk -V
Asterisk 1.8.2.3
2.)dahdi
*CLI> dahdi show version
DAHDI Version: 2.4.0 Echo Canceller: MG2
3.) lipri
*CLI> pri show version
libpri version: 1.4.11.5
I've already tried to call over each channel from 1 to 15 (i have only
15B channels)
exten => _X.,n,Dial(DAHDI/1/${EXTEN})
exten => _X.,n,Dial(DAHDI/2/${EXTEN})
....
exten => _X.,n,Dial(DAHDI/15/${EXTEN})
but everytime i am getting the same DIALSTATUS
<snip>
-- Channel 0/1, span 1 got hangup request, cause 31
...
-- Auto fallthrough, channel 'SIP/2000-00000002' status is 'CHANUNAVAIL'
</snip>
Regards,
Robert
On 21.02.2011 12:13, Andrew Thomas wrote:
> I'm curious as to what versions of everything you are using. Reason
> being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing
> it to SIP/5000-00000000".
>
> It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that
> before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to
> SIP/801-0000000c" [1-1 being the span and channel numbers]).
>
> What happens if you change "exten => _X.,n,Dial(DAHDI/g1/${EXTEN})" to
> "exten => _X.,n,Dial(DAHDI/1/${EXTEN})"?
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