ZAP Dialling Error

[uxbod]

Hi,

I am running Asterisk 1.4.1 and am having problems with outbound dialling via zap. I get the following error when trying :-

-- Executing [9674035@internal-nonuri:1] Dial("SIP/uxbod-00666190", "Zap/g0/123456") in new stack [Mar 3 18:00:05] WARNING[26731]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [9123456@internal-nonuri:2] Congestion("SIP/uxbod-00666190", "") in new stack == Spawn extension (internal-nonuri, 9123456, 2) exited non-zero on 'SIP/uxbod-00666190'

Dial(${PSTN}${EXTEN:1})

PSTN has been set too Zap/g0 which is my FXO module on a TDM11B.

Incoming works fine.

Here is my zapata.conf :-

[code][channels]
;
language=en
context=incoming
switchtype=national
sendcalleridafter=2

signalling=fxo_ks
context=internal
channel => 1
callerid=“Office” <1000>
mailbox=1000
group=5

signalling=fxs_ks
context=incoming
channel => 4
callerid=asreceived
group=0
[/code]

Any ideas ?

Note: Real number has been replaced with 123456!

[alunt2003]

Try the following in extensions.conf:

[code]Under your [internal] section put:

include => outbound

Remove your Dial(${PSTN}${EXTEN:1}) then add a section like this:

[outbound]
exten => _0XXXXXXXXXX,1,Dial(Zap/4/${EXTEN})
exten => _0XXXXXXXXXX,2,Congestion()
exten => _0XXXXXXXXXX,102,Congestion()
[/code]

[uxbod]

Tried and get a engaged tone yet the line is available :-

Connected to Asterisk 1.4.1 currently running on mailhub (pid = 28666) Verbosity was 0 and is now 4 -- Starting simple switch on 'Zap/1-1' -- Executing [9123456@internal:1] GotoIf("Zap/1-1", "1?6") in new stack -- Goto (internal,9123456,6) -- Executing [9123456@internal:6] Goto("Zap/1-1", "internal-nonuri|9123456|1") in new stack -- Goto (internal-nonuri,9674035,1) -- Executing [91234565@internal-nonuri:1] Dial("Zap/1-1", "Zap/4/123456") in new stack -- Called 4/123456 -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [9123456@internal-nonuri:2] Congestion("Zap/1-1", "") in new stack == Spawn extension (internal-nonuri, 9123456, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1'

[alunt2003]

You have “GotoIf” and “Goto” in your output which tells me you still have a dialplan with old stuff in it.

Remove all the old stuff and do an “extensions reload” in the asterisk cli or restart asterisk

[ianplain]

Hi Uxbod

Could you post the actual section of the extensions.conf.
The dbug shows us whats happening but we need to see what you are expecting to happen.

Ian

[uxbod]

[code][general]
;
static=yes
writeprotect=no
clearglobalvars=no
;
[globals]

CONSOLE=Console/dsp ; Console interface for demo
PSTN=Zap/g0 ; Trunk interface
OFFICE=Zap/1
FQDN=mailhub.splatnix.net

[incoming]

exten => s,1,GoToIfTime(17:00-09:00|mon-fri||?outofhours,s,1)
exten => s,2,GoToIfTime(|sat-sun||*?outofhours,s,1)
exten => s,3,Answer()
exten => s,4,Dial(Zap/1,20)
exten => s,5,Playback(cybermog)
exten => s,6,VoiceMail(1000@incoming,s)
exten => s,7,Hangup()
[outofhours]
exten => s,1,Answer()
exten => s,2,Playback(cybermog)
exten => s,3,VoiceMail(1000@incoming,s)
exten => s,4,Hangup()

[internal]
exten => _.,1,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?6)
exten => _.,2,GotoIf($[${SIPDOMAIN} = ${FQDN}]?6)
exten => _.,3,GotoIf($[${SIPDOMAIN} = ${FQDN}:5060]?6)
exten => _.,4,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _.,5,HangUp()
exten => _.,6,Goto(internal-nonuri,${EXTEN},1)
exten => h,1,HangUp()

[internal-nonuri]
include => internal-extensions
include => outbound-local
include => outbound-national

[internal-extensions]
exten => 5000,1,VoiceMailMain(@incoming)
exten => 5000,2,Hangup()
exten => 1000,1,Dial(${OFFICE},20)
exten => 1000,2,VoiceMail(1000@incoming)
exten => 1000,3,Hangup()
exten => 1000,101,VoiceMail(1000@incoming)
exten => 1000,102,Hangup()
exten => 1001,1,Dial(sip/uxbod,20)
exten => 1001,2,VoiceMail(1001@incoming,u)
exten => 1001,3,Hangup()
exten => 1001,101,VoiceMail(1001@incoming,u)
exten => 1001,102,Hangup()

[outbound-local]
exten => _9NXXXXX,1,Dial(${PSTN}/${EXTEN:1})
exten => _9NXXXXX,2,Congestion()
exten => _9NXXXXX,102,Congestion()

[outbound-national]
exten => _9XXXXXNXXXXX,1,Dial(${PSTN}/${EXTEN:1})
exten => _9XXXXXNXXXXX,2,Congestion()
exten => _9XXXXXNXXXXX,102,Congestion()

[macro-uridial]
exten => s,1,Dial(SIP/${ARG1})
exten => s,2,Congestion()

[inbound-sip]
exten => uxbod,1,Dial(sip/uxbod,20)
exten => uxbod,2,VoiceMail(1001@incoming)
exten => uxbod,3,Hangup()
[/code]

[baconbuttie]

you should specify a group= in zapata.conf before the channel= line.

[alunt2003]

Can you change:

PSTN=Zap/g0 to PSTN=Zap/4

reload and see what happens?

In my box i have Zap/4 as the connection to my phone line, and Zap/1 as the connection to my analogue phone.

I don’t see why yours would be Zap/g0 ( but then i am a semi-newbie)

Just trying to help

[ianplain]

Hi

Basicly you havent defined group zero.

In zapata.conf seting are inherited from above unless specified.

Try

[code][channels]
;
language=en
sendcalleridafter=2

signalling=fxo_ks
context=internal
callerid=“Office” <1000>
mailbox=1000
group=5
channel => 1

signalling=fxs_ks
context=incoming
callerid=asreceived
group=0
channel => 4

[quote]

[/quote][/code]

[uxbod]

Ian,

If you look at my original posting group 0 is defined in zapata.conf for the FXO module. But even if I dont use Zap/go and just use Dial(Zap/4/${EXTEN}) that doesn’t work neither

[uxbod]

Hmmm, I reckon it is something in 1.4.X release. Will drop back too 1.2.15 as it has worked in that release fine.

[alunt2003]

I have the same setup as you.

Asterisk 1.4
Zaptel 1.4

I have my PSTN line on channel=4
I have my analogue phone on channel=1

I have group=1 above channel=1
I have group=2 above channel=4

All my dialing out goes on Zap/4

Have you tried makeing a simple extension like so:

exten => 1471,1,Dial(Zap/4/1471,10,r) exten => 1471,2,Hangup()

Or try changing Zap/4 to Zap/g0

Why is your PSTN line Zap/g0 and mine is Zap/4 do you know?

[uxbod]

Zap/g0 should hunt for group 0 and find the first line available, if you have more than one PSTN, but I have also tried with Zap/4/ on a extension and that fails aswell

[alunt2003]

So the PSTN line is definately on channel 4 then, (physically)

Looking at the card, is the Green module on channel 1 & the Red module on channel 4?

I presume you are already sure but sometimes it’s a simple mistake that can baffle us for days.

[ianplain]

Hi

Whats in zaptel.conf?

What does zap show channels say ?

also what is shown in dmesg.

Ian

[uxbod]

Yes the modules are correct, as if I dial a extension that goes to Zap/1 it works fine I am just trying release 1.2.16 as it has worked before. Only seems to fail on 1.4.1. Just wondering if it is to do with the fact that in UK.

[alunt2003]

Nah, I’m in the UK using asterisk 1.4 & zaptel 1.4 as well

[uxbod]

Okay, I have just deleted everything and have reinstalled. Damn things wants to work now

Here is my zapata.conf :-

[code][channels]
context=internal

signalling=fxo_ks
callerid=“Office” <1000>
mailbox=1000
group=5
context=internal
channel => 1
callerid=
mailbox=
group=0

;;; line="4 WCTDM/0/3 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
cidsignalling=v23
cidstart=polarity
context=incoming
channel => 4
[/code]

One thing I did notice though when re-installing was that with the default samples if you ran genzaptelconf it does not find zapata-channels.conf.

Thanks very much for everyones help though.

[alunt2003]

Glad it works now.

On a side note, do you get callerid on your analogue phone?
For me, Asterisk sees the callerid and passes it to my sip phone, but not my analgue phone

[uxbod]

Just tried it and no it does not get passed to the analogue extension