asterisk users - Feb 2011 - calls are not going thru e1 line

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})"?



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Albert
Sent: 17 February 2011 16:56
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] calls are not going thru e1 line


On 17.02.2011 17:47, Danny Nicholas wrote: 
<snip>
Post your "dahdi show channels" output.

Have you checked the lines with a regular handset?

here it is:
*CLI> dahdi show status
Description                              Alarms  IRQ    bpviol CRC4
Fra Codi Options  LBO
T2XXP (PCI) Card 0 Span 1                OK      0      0      0
CCS HDB3 CRC4     0 db (CSU)/0-133 feet (DSX-1)
T2XXP (PCI) Card 0 Span 2                UNCONFI 0      0      0
CAS Unk           0 db (CSU)/0-133 feet (DSX-1)

*CLI> dahdi show channels
   Chan Extension  Context         Language   MOH Interpret
Blocked    State     
 pseudo            default                    default
In Service
      1            incoming_calls  en         default
In Service
      2            incoming_calls  en         default
In Service
      3            incoming_calls  en         default
In Service
      4            incoming_calls  en         default
In Service
      5            incoming_calls  en         default
In Service
      6            incoming_calls  en         default
In Service
      7            incoming_calls  en         default
In Service
      8            incoming_calls  en         default
In Service
      9            incoming_calls  en         default
In Service
     10            incoming_calls  en         default
In Service
     11            incoming_calls  en         default
In Service
     12            incoming_calls  en         default
In Service
     13            incoming_calls  en         default
In Service
     14            incoming_calls  en         default
In Service
     15            incoming_calls  en         default
In Service
     17            incoming_calls  en         default
In Service
     18            incoming_calls  en         default
In Service
     19            incoming_calls  en         default
In Service
     20            incoming_calls  en         default
In Service
     21            incoming_calls  en         default
In Service
     22            incoming_calls  en         default
In Service
     23            incoming_calls  en         default
In Service
     24            incoming_calls  en         default
In Service
     25            incoming_calls  en         default
In Service
     26            incoming_calls  en         default
In Service
     27            incoming_calls  en         default
In Service
     28            incoming_calls  en         default
In Service
     29            incoming_calls  en         default
In Service
     30            incoming_calls  en         default
In Service
     31            incoming_calls  en         default
In Service

Yes, is was checked and calls were going through line.

Regards,
Albert


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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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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})"?


 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.