[Asterisk-Users] Can't call out on ZAP channel - need help
O'Connor, Jonathan
Jonathan.OConnor at inoveris.com
Mon Dec 19 08:29:02 MST 2005
The parameter in zaptel.conf that sets up timing etc is:
span=1,1,0,esf,b8zs
The first 1 means this is span 1. The second one defines the timing of
the link. For asterisk to provide the timing use 0 instead. For
instance my Asterisk box, hooked directly to my Avaya G3 uses:
span=1,0,0,esf,b8zs
Also,
switchtype=qsig
This is something I have never personally got working to any useful
amount with our Definity. I use
switchtype=national
It doesnt have some of the features of qsig, but will get you going if
the PBX is setup to use a standard National ISDN 2 switch.
You will I beleive need to shut down asterisk and then run ztcfg -vvvv
if you make these changes, then restart asterisk.
signalling=pri_net merely makes the Asterisk box act like the telco, as
far as its signaling is concerned, quite normal when hooked to a legecy
pbx.
Hope this helps, am no expert, just going on what I got mine running
with :)
-Jonathan
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael
Sampson
Sent: Monday, December 19, 2005 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can't call out on ZAP channel -
need help
Yeah, the zttool program shows the PRI as having No Alarms. It
is an Infinity system by Amtelco. I haven't actually tried making a call
from the other pbx, but I did have my vendor (Amtelco) look at it and
they verified that the span was up and everything was working correctly.
The asterisk system is set to signalling=pri_net which I assumed meant
that the asterisk box would be handling the timing.
Here is the output from "pri show span 1"
asterisk1*CLI> pri show span 1
Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: Q.SIG switch
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T313 Timer: 4000
Michael Sampson
Information Systems Manager
Customer Contact Services
msampson at yourccsteam.com
952-936-4000
O'Connor, Jonathan wrote:
Michael,
Does the zttool program show the PRI as working
correctly?
Can the PBX push calls into the Asterisk system?
Also, what type of PBX is it, and is it providing the
clock etc.. For
the T1 connection?
-Jonathan
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]
On Behalf Of
Michael Sampson
Sent: Monday, December 19, 2005 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Can't call out on ZAP
channel - need help
I'm trying to connect to another PBX via an T-1
interface. I
have a T100P card.
On the CLI I get the error "Everyone is
busy/congested at
this time (1:0/0/1)" When I try to dial out of
the T-1 line
from an SIP softphone.
I have posted this question a few times here and
at the
asterisk forum, but can't get anyone to respond.
I've seen
other people on forums with the same problem but
no one has
ever given much of a solution. Does someone at
least know
what the next step in debugging this problem
would be.
In the file /var/log/asterisk/full I get the
error "Unable to
create channel of type 'ZAP'"
Here are my configs.
Zapata.conf
------------------
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
;signalling=fxs_ks
signalling=pri_net ; pri_cpe= PRI slave ;
pri_net = PRI
master switchtype=qsig pridialplan=local
resetinterval=never
;rxwink=300 ; Atlas seems to use long
(250ms) winks
;
; Whether or not to do distinctive ring
detection on FXO
lines ; ;usedistinctiveringdetection=yes
callerid=asreceived
usecallerid=yes hidecallerid=no callwaiting=yes
usecallingpres=yes callwaitingcallerid=yes
threewaycalling=yes transfer=yes
cancallforward=yes
callreturn=yes echocancel=yes
echocancelwhenbridged=yes
echotraining=400 rxgain=0.0 txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include genzaptelconf configs
#include zapata-auto.conf
;Include AMP configs
#include zapata_additional.conf
channel => 1-23
-------------------------
Zaptel.conf
-------------------------
# Autogenerated by /usr/local/sbin/genzaptelconf
-- do not
hand edit # Zaptel Configuration File # # This
file is parsed
by the Zaptel Configurator, ztcfg #
# It must be in the module loading order
# Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI
Card 0"
# channel 1, WCT1, unhandled for now
# channel 2, WCT1, unhandled for now
# channel 3, WCT1, unhandled for now
# channel 4, WCT1, unhandled for now
# channel 5, WCT1, unhandled for now
# channel 6, WCT1, unhandled for now
# channel 7, WCT1, unhandled for now
# channel 8, WCT1, unhandled for now
# channel 9, WCT1, unhandled for now
# channel 10, WCT1, unhandled for now
# channel 11, WCT1, unhandled for now
# channel 12, WCT1, unhandled for now
# channel 13, WCT1, unhandled for now
# channel 14, WCT1, unhandled for now
# channel 15, WCT1, unhandled for now
# channel 16, WCT1, unhandled for now
# channel 17, WCT1, unhandled for now
# channel 18, WCT1, unhandled for now
# channel 19, WCT1, unhandled for now
# channel 20, WCT1, unhandled for now
# channel 21, WCT1, unhandled for now
# channel 22, WCT1, unhandled for now
# channel 23, WCT1, unhandled for now
# channel 24, WCT1, unhandled for now
# Global data
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#fxsks=1
loadzone = us
defaultzone = us
-----------------------------
--
Michael Sampson
Information Systems Manager
Customer Contact Services
msampson at yourccsteam.com
952-936-4000
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