[Asterisk-Users] TDM

[Curt Shaffer]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 *@H it is asterisk with FreePBX only. Yes the phone line is
> connected to the right port. No luck. Thanks.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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 Asterisk at home 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 
>>>       
>>     
>
>   

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3115 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060528/44bb6ee2/smime.bin