[asterisk-users] calls are not going thru e1 line

Tue Feb 22 17:09:34 CST 2011

Hi Andrew,

thanks for your answer. I haven't notice this typo before, i was
replacing this config sooooo many times ;-)

I did as you suggested, replaced with your config but result is still
the same.

Some technicians from telco came yesterday to investigate and confirmed
that something is wrong at they end, now i am waiting for them to clear
this issue.

I am not setting this up in UK, but in Uganda. That's why i am using
loadzone from UK.

I will keep you posted if my issue was solved.

Thanks,
Albert

On 22.02.2011 11:55, Andrew Thomas wrote:
> This is very strange.  Everything matches mine except Asterisk itself
> (I'm using 1.6.2.16.1).
>
> I did notice that you set the loadzone(s) for UK use - yet your e-mail
> address in in Poland.  Are you setting this up in the UK?
>
> BTW - you have a typo in chan_dahdi.conf ("busydetec=yes" is missing the
> 't' [I wonder if this is causing your problem - as the 'include' is
> after this]) and I'd cetainly remove "pulsedial=yes" ;).
>
> Anyway, here's the part of my chan_dahdi.conf that is working for me
> (I've changed the context to match yours):
>
> ;chan_dahdi.conf
>
> [trunkgroups]
>
> [channels]
> language = en
> context = incoming_calls
> switchtype = euroisdn
> pridialplan = unknown
> prilocaldialplan = unknown
> internationalprefix = 00
> nationalprefix = 0
> localprefix =
> unknownprefix =
> rxwink = 300
> usecallerid = yes
> hidecallerid = no
> callwaiting = yes
> usecallingpres = yes
> sendcalleridafter = 1
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> rxgain = 0.0
> txgain = 0.0
> group = 1
> callgroup = 1
> pickupgroup = 1
> immediate = no
> faxdetect = no
> echocancel = yes
> echocancelwhenbridged = no
> echotraining = yes
> signalling = pri_cpe
> channel => 1-15,17-31
>
> Maybe drop mine in as a replacement and see what happens then (remember
> to back yours up).
>
> BTW - you don't need to include dahdi-channels.conf in the above - as
> it's already included.
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Albert
> Sent: 21 February 2011 13:53
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] calls are not going thru e1 line
>
>
> 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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