[Asterisk-Users] TDM

[Steve Totaro]

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: 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 
>>>>       
>>>>         
>>>     
>>>       
>>   
>>     
>
>
>