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Hi Andrew,<br>
<br>
thanks for your answer. I haven't notice this typo before, i was
replacing this config sooooo many times ;-)<br>
<br>
I did as you suggested, replaced with your config but result is
still the same.<br>
<br>
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.<br>
<br>
I am not setting this up in UK, but in Uganda. That's why i am using
loadzone from UK.<br>
<br>
I will keep you posted if my issue was solved.<br>
<br>
Thanks,<br>
Albert<br>
<br>
<br>
On 22.02.2011 11:55, Andrew Thomas wrote:
<blockquote
cite="mid:7EA66C795632264A8E49BCFF2DCDF1FC22710A@server.DataVox.local"
type="cite">
<pre wrap="">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: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Albert
Sent: 21 February 2011 13:53
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
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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</pre>
</blockquote>
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