It looks OK. Try editing extensions.conf and add an extension in a context that will included when you dial.

Try something like this
exten => 123,1,Dial(ZAP/g0/1NXXNXXXXXX)

The open the console and dial 123.

This will bypass any funky dialplan issues with FreePBX. If it works, then obviously something is not right in FreePBX. If it doesnt' then that indicates your configuration files need tweaking.

Thanks,
Steve

Curt Shaffer wrote:
Here is the output from a dial when starting asterisk with -vvvvv. The
1NXXNXXXXXX is actually the number not those characters FYI.

Thanks

-- Executing Macro("SIP/103-a555", "dialout-trunk|1|1NXXNXXXXXX||") in new
stack
    -- Executing GotoIf("SIP/103-a555", "1?3:2") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/103-a555", "user-callerid") in new stack
    -- Executing GotoIf("SIP/103-a555", "0?report") in new stack
    -- Executing GotoIf("SIP/103-a555", "0?start") in new stack
    -- Executing Set("SIP/103-a555", "REALCALLERIDNUM=103") in new stack
    -- Executing NoOp("SIP/103-a555", "REALCALLERIDNUM is 103") in new stack
    -- Executing Set("SIP/103-a555", "AMPUSER=103") in new stack
    -- Executing Set("SIP/103-a555", "AMPUSERCIDNAME=103") in new stack
    -- Executing GotoIf("SIP/103-a555", "0?report") in new stack
    -- Executing Set("SIP/103-a555", "CALLERID(all)=103 <103>") in new stack
    -- Executing NoOp("SIP/103-a555", "Using CallerID "103" <103>") in new
stack
    -- Executing Macro("SIP/103-a555", "record-enable|103|OUT") in new stack
    -- Executing GotoIf("SIP/103-a555", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/103-a555",
"recordingcheck|20060528-110627|1148832387.1") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060528-110627|1148832387.1: Outbound recording not
enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/103-a555", "No recording needed") in new stack
    -- Executing Macro("SIP/103-a555", "outbound-callerid|1") in new stack
    -- Executing GotoIf("SIP/103-a555", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/103-a555", "REALCALLERIDNUM is 103") in new stack
    -- Executing Set("SIP/103-a555", "USEROUTCID=") in new stack
    -- Executing Set("SIP/103-a555", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/103-a555", "TRUNKOUTCID=") in new stack
    -- Executing GotoIf("SIP/103-a555", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing GotoIf("SIP/103-a555", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,13)
    -- Executing GotoIf("SIP/103-a555", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing NoOp("SIP/103-a555", "CallerID set to "103" <103>") in new
stack
    -- Executing Set("SIP/103-a555", "GROUP()=OUT_1") in new stack
    -- Executing GotoIf("SIP/103-a555", "0?108") in new stack
    -- Executing Set("SIP/103-a555", "DIAL_NUMBER=1NXXNXXXXXX") in new stack
    -- Executing Set("SIP/103-a555", "DIAL_TRUNK=1") in new stack
    -- Executing AGI("SIP/103-a555", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/103-a555", "OUTNUM=1NXXNXXXXXX") in new stack
    -- Executing Set("SIP/103-a555", "custom=ZAP/g0") in new stack
    -- Executing GotoIf("SIP/103-a555", "0?16") in new stack
    -- Executing Dial("SIP/103-a555", "ZAP/g0/1NXXNXXXXXX|120|r") in new
stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Goto("SIP/103-a555", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing NoOp("SIP/103-a555", "Dial failed due to CHANUNAVAIL") in
new stack
    -- Executing Macro("SIP/103-a555", "outisbusy|") in new stack
    -- Executing Playback("SIP/103-a555", "all-circuits-busy-now") in new
stack
    -- Playing 'all-circuits-busy-now' (language 'en')
    -- Executing Playback("SIP/103-a555", "pls-try-call-later") in new stack
    -- Playing 'pls-try-call-later' (language 'en')
  == Spawn extension (macro-outisbusy, s, 2) exited non-zero on
'SIP/103-a555' in macro 'outisbusy'
  == Spawn extension (macro-outisbusy, s, 2) exited non-zero on
'SIP/103-a555'

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Sunday, May 28, 2006 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM

Connect to the Asterisk console with verbose turned on and try to dial. Post that output.
Curt Shaffer wrote:
This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone 
line is
connected to the right port. No luck. Thanks.

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Saturday, May 27, 2006 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM



Steve Totaro wrote:

Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm?

Would he get the ztcfg message if it were not?
Is the phone line plugged into the correct jack?
With only one module installed, the other three jacks lead to nowhere.
Also this seems to be [EMAIL PROTECTED] from the references, so perhaps there is a context issue that the configuration files address.
AAH can really lead one down the garden path!

John Novack

Curt Shaffer wrote:

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card.

Here is my /etc/zaptel.conf

loadzone=us

fxsks=1

and here is my /etc/Zapata.conf

[channels]

language=en

#include zapata_additional.conf

context=from-zaptel

signalling=fxs_ks

faxdetect=incoming

usecallerid=asreceived

echocancel=yes

callprogress=no

busydetect=no

echocancelwhenbridged=no

echotraining=800

group=0

channel=>1

When I dial in Asterisk does not even show an initiation of the call. When I dial out on that trunk I get all circuits busy. Ztcfg -vvv shows the following

ztcfg -vvv

Zaptel Configuration

======================

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

Any help would be appreciated.

Curt



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