[asterisk-users] US T1 Hangup Detection

Matt Florell astmattf at gmail.com
Tue Jul 8 12:17:51 CDT 2008


Is there any way you could get a cut-sheet from Verizon. I know they
are difficult to work with, but it would help to see for sure if your
circuit is indeed Loop-start. You could always try E&M_wink or E&M
immediate and see if there is any change.

MATT---

On 7/8/08, Daniel Hazelbaker <daniel at highdesertchurch.com> wrote:
> > 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
>
>
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