[Asterisk-Users] Can't call out on ZAP channel - need help
Michael Sampson
msampson at yourccsteam.com
Mon Dec 19 09:04:33 MST 2005
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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>>>
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