[asterisk-users] US T1 Hangup Detection

Daniel Hazelbaker daniel at highdesertchurch.com
Tue Jul 8 12:00:08 CDT 2008


> Date: Mon, 7 Jul 2008 16:48:00 -0400
> From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
>
> Digital ISDN used Q931 messages.  You should get a disconnect message
> from telco on the d-channel 23.

I am pretty sure it is a T1 and not a PRI.  I did try configuring it  
as a PRI and it started spewing all kinds of errors and completely  
stopped working.


> Date: Mon, 07 Jul 2008 16:55:27 -0400
> From: Doug Lytle <support at drdos.info>
>
> Daniel Hazelbaker wrote:
>> We are in the process of preparing to move our Asterisk server to a
>> Digital T1 interface card instead of a analog card (via an Adtran
>> which is now connected to the T1).  I did a preliminary test the  
>> other
>>
>
> A T1 or a PRI?  Just make sure we're on the same page.
> Also, show us your zaptel and zapata.conf


Again, I am pretty sure T1.  It is a Verizon "Flex-Grow" package,  
which they list as expandable up to 24 voice channels.  That and I  
tried configuring as a PRI and it harfed.  The Adtran box we use now  
is configured as:

Timing Mode     Network
Format          ESF
Line Code       B8ZS
Equalization    0 dB
CSU Lpbk        Enable
Rx Sensitivity  Auto

Right now with Asterisk "mostly" working (it answers calls, dials out,  
etc. just doesn't detect hangup) my /etc/zaptel.conf is:
#
# Span Configuration
# ~~~~~~~~~~~~~~~~~~
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs

#
# Channel Configuration
# ~~~~~~~~~~~~~~~~~~~~~
fxsks=1-24
fxoks=25-48

loadzone = us
defaultzone=us
--CUT--

/etc/asterisk/zapata.conf:
[channels]
usecallerid=yes
callerid=asreceived
cidsignalling=bell
cidstart=ring
callprogress=yes			# I have turned this off too

;-------------------------------------------------
;
; Define telco channels in rotary, these should be answered
; like a normal incoming call.
;
context=bridgeNEC
usecallerid=yes
signalling=fxs_ks
group=1                 ; Part of ZAP group 1
channel => 1-9

context=incoming
channel => 12

;-------------------------------------------------
;
; Telco line, computer dialup, needs to be routed to output line.
;
group=2
usecallerid=no
channel => 10           ; PSTN attached to Span1:Port10

;-------------------------------------------------
;
; Telco line, construction trailer fax, needs to be routed.
;
group=3
usecallerid=no
channel => 11           ; PSTN attached to Span1:Port11


;-------------------------------------------------
;
; ADTran lines, used for outgoing to analog devices
;
context=incoming
group=4
usecallerid=no
signalling=fxo_ks
channel => 25-36
--CUT--

For context, the bridgeNEC context just dials out one of the ADTran  
lines to our existing NEC system, but the incoming context starts our  
menu-system, which was also not detecting hangups.

I have also tried using loopstart and groundstart signalling, doesn't  
seem to make a difference.  I am pretty well stumped myself.  I need  
to call the telco about the caller id not working to verify that it is  
still turned on, but I figure I might as well wait so that if I need  
to ask them about the signalling I can know all the questions to ask  
at the same time.

>
Thanks,
Daniel



More information about the asterisk-users mailing list