asterisk users - Feb 2006 - problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)

Nik,

Looks like you're making some progress.  When I first started using A@H
I had trouble getting the outbound dialing to work.  I wasn't sure where
to start, so what I did was skip the macros in the dial plan.  I wanted
to play around with exactly what digits the telco wanted to see.  So I
put a specific extension in my [default] context like this:

exten => 555,1,Dial(Zap/1/5595551212)

I chose a specific Zap channel and the exact digits that I wanted to
send to the telephone company.  This helped me figure out what to dial.

The other thing you can do is log on to the CLI and turn on PRI
debugging:

pri debug span 1

This will cause PRI debug messages to display on the console.  It might
take a while but you will learn to read those debug messages.  You can
also post them to the list and we'll help you to interpret them.

-MC

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of nik600
Sent: Wednesday, February 15, 2006 1:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problem with outgoing
callsUnabletocreatechannel of type 'ZAP' (cause 34 -
Circuit/channelcongestion)

On 2/14/06, Michael Collins <mcollins@fcnetwork.biz>
wrote:
> Nik,
>
> I'm not sure that "NOP" is correct, but I'm in the states
so I'll to
> defer to someone who knows E1/PRI.  When I run zttool I have "OK"
under
> the alarms.  Is there a way you can call the telco and confirm the
> settings?  Make sure that framing, coding and D channels are set up on
> their end the same way you're set up.
>
ok, with your configuration incoming calls works, but:

- i have eco (maybe i have to increase/decrease echotraining value?)
- outgoing calls doesn't works (

    -- Executing Dial("SIP/102-cc9b", "ZAP/g0/mynumber") in
new stack
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/102-cc9b", "s-CONGESTION|1") in
new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing NoOp("SIP/102-cc9b", "Dial failed due to
CONGESTION")
in new stack
    -- Executing Macro("SIP/102-cc9b", "outisbusy") in new
stack
    -- Executing Playback("SIP/102-cc9b",
"all-circuits-busy-now") in
new stack

)
it seems that i don't have any channel for outbound
- ALARM is set on NOP

i've got a TE205P and my zaptel.conf is:

span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow

bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47

loadzone        = it
defaultzone     = it
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On 2/15/06, Michael Collins <mcollins@fcnetwork.biz>
wrote:
> Nik,
>
> Looks like you're making some progress.  When I first started using A@H
> I had trouble getting the outbound dialing to work.  I wasn't sure
where
> to start, so what I did was skip the macros in the dial plan.  I wanted
> to play around with exactly what digits the telco wanted to see.  So I
> put a specific extension in my [default] context like this:
>
> exten => 555,1,Dial(Zap/1/5595551212)
>
> I chose a specific Zap channel and the exact digits that I wanted to
> send to the telephone company.  This helped me figure out what to dial.
>
> The other thing you can do is log on to the CLI and turn on PRI
> debugging:
>
> pri debug span 1
>
> This will cause PRI debug messages to display on the console.  It might
> take a while but you will learn to read those debug messages.  You can
> also post them to the list and we'll help you to interpret them.
>
> -MC
ok, thanks for your support, now i've enabled debug on span 1, and
i've make a new entry in extension.conf:

exten => 444,1,Dial(Zap/0/mynumber)

when i call 444 i get in the logs:

Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Setting NAT on RTP to 0
Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Checking SIP call limits for device 102
Feb 17 03:50:59 DEBUG[3607] chan_sip.c: build_route: Contact hop:
<sip:102@192.168.100.180:5060>
Feb 17 03:50:59 VERBOSE[4262] logger.c:     -- Executing
Dial("SIP/102-2079", "Zap/0/mynumber") in new stack
Feb 17 03:50:59 NOTICE[4262] app_dial.c: Unable to create channel of
type 'Zap' (cause 0 - Unknown)
Feb 17 03:50:59 VERBOSE[4262] logger.c:   == Everyone is
busy/congested at this time (1:0/0/1)
Feb 17 03:50:59 DEBUG[4262] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

it seems that the only information it gives mi is:

 app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

so it seems that i don't have channel for outgoing calls? how can i check
it?
maybe there is another logfile more detailed?

thanks a lot for your help...