[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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