The only other thing I can think of is that your contexts etc... need checked. 
 
It would be very helpful to know if calls can come into the system from the PBX, that would be the only way to know the span is alive and well truely.  Once you know that then its down to the contexts and configs...
 
 
 
 
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
 
 
 


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Sampson
Sent: Monday, December 19, 2005 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can't call out on ZAP channel - need help

My other pbx vendor told me they supported pretty much all of the switchtypes and that the system would automatically detect the correct one. I've tried Qsig and National and both seem to bring the span up fine.

I just switched to
span=1,0,0,esf,b8zs to have asterisk provide the timing. That didn't change any of the errors I'm getting. So I changed the switchtype to national just to be sure, and it still didn't fix anything. Everything seems to indicate that the span is up and running fine.

Any more ideas?
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000


O'Connor, Jonathan wrote:
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] 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
[EMAIL PROTECTED]
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: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] 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
[EMAIL PROTECTED]
952-936-4000

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