[Asterisk-Users] Can't call out on ZAP channel - need help

O'Connor, Jonathan Jonathan.OConnor at inoveris.com
Mon Dec 19 09:13:27 MST 2005


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: 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 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
	msampson at yourccsteam.com
	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: 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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