Help with TDM410 Dialout!

[v0ip-Pete]

Hi,

I have system setup so that normal calls are dialed via IAX trunk but calls preceded with 9 dialout via ZAP.
I have TDM410 with 2 x FXS (green) modules connected to PSTN. Pbxiaf is all set up, but when I dial I get ringing tone on the phone but it's not actuall dialling out.

Last messages I get are

called g1/xxxxxxx
Zap/3-1 is ringing...

Card is detected ok, I have run the config tool.

Any suggestions?

Thanks

 

[v0ip-Pete]

Anyone?

 

[Calunker]

FXS (green) modules are for terminating analog telephones. You need a red FXO module to terminate a PSTN line.

 

[Bart]

Not enough info and guesses won't help you. Ringing is fake and means nothing to you.

More CLI output might help. How are your trunks setup in zapata-auto? What group?

Bart

 

[v0ip-Pete]

Hope I haven't bought the wrong modules, can't think why anyone would want to hook a normal phone up to the system.

CLI output:

-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/2003-085cc508", "OUTNUM=xxxxxxxxxxxxxxx6") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/2003-085cc508", "custom=ZAP/g1") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/2003-085cc508", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/2003-085cc508", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2003-085cc508", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/2003-085cc508", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/2003-085cc508", "ZAP/g1/xxxxxxxxxxxxxx|300|") in new stack
-- Called g1/xxxxxxxxxxx
-- Zap/3-1 is ringing
-- Zap/3-1 is ringing
-- Zap/3-1 is ringing


-----------------------------------------------------------

zapata-auto.conf is


callerid=asreceived

; Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER)
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1
context=from-zaptel
group=0
channel => 1

signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 2
context=from-zaptel
group=0
channel => 2

signalling=fxo_ks
; Note: this is an extension. Create a ZAP extension in AMP for Channel 3
context=from-internal
group=1
channel => 3

signalling=fxo_ks
; Note: this is an extension. Create a ZAP extension in AMP for Channel 4
context=from-internal
group=1
channel => 4



Do this make sense?

Thanks

 

[jwells]

My understanding is if you connect a FXS port to an outside line and then a call comes in the ring voltage will fry the FXS port as well. I would remove them asap if you have not already blown them. The FX0 module is designed to handle the ring voltage.

 

[Bart]

According to your zappata-auto, you have 4 modules. First two are your Phone Lines, Next two are extensions.
If you do, it appears you have g1 in trunks setting, not g0. Change that and see if you have better luck.
If you don't, then you'll need to purchase FXO modules for Telco Lines

Bart

 

[v0ip-Pete]

Hi, Changed trunk setting to g0

Get no ringing now!

-- Executing [90xxxxxxxxx@from-internal:4] Macro("SIP/2003-08589910", "record-enable|2003|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/2003-08589910", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/2003-08589910", "recordingcheck|20080709-002139|1215559299.79") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080709-002139|1215559299.79: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/2003-08589910", "No recording needed") in new stack
-- Executing [90xxxxxxxxx@from-internal:5] Macro("SIP/2003-08589910", "dialout-trunk|1|0xxxxxxxxxxx||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/2003-08589910", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/2003-08589910", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2003-08589910", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/2003-08589910", "DIAL_NUMBER=0xxxxxxxx") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/2003-08589910", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/2003-08589910", "GROUP()=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2003-08589910", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2003-08589910", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/2003-08589910", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/2003-08589910", "outbound-callerid|1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2003-08589910", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] GotoIf("SIP/2003-08589910", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing [s@macro-outbound-callerid:4] NoOp("SIP/2003-08589910", "REALCALLERIDNUM is 2003") in new stack
-- Executing [s@macro-outbound-callerid:5] GotoIf("SIP/2003-08589910", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,10)
-- Executing [s@macro-outbound-callerid:10] Set("SIP/2003-08589910", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/2003-08589910", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:12] Set("SIP/2003-08589910", "TRUNKOUTCID=0xxxxxxxxxxx") in new stack
-- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/2003-08589910", "0?trunkcid") in new stack
-- Executing [s@macro-outbound-callerid:14] GotoIf("SIP/2003-08589910", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,17)
-- Executing [s@macro-outbound-callerid:17] GotoIf("SIP/2003-08589910", "0?usercid") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/2003-08589910", "CALLERID(all)=0xxxxxxxxx") in new stack
-- Executing [s@macro-outbound-callerid:19] GotoIf("SIP/2003-08589910", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,23)
-- Executing [s@macro-outbound-callerid:23] NoOp("SIP/2003-08589910", "CallerID set to "" <0xxxxxxxxxx>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/2003-08589910", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern .
== fixlocalprefix: Dialpattern . matched. 0xxxxxxxx -> 0xxxxxxxxx
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/2003-08589910", "OUTNUM=0xxxxxxxx") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/2003-08589910", "custom=ZAP/g0") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/2003-08589910", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/2003-08589910", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2003-08589910", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/2003-08589910", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/2003-08589910", "ZAP/g0/0xxxxxxx|300|") in new stack
-- Called g0/0xxxxxxxx
-- Zap/1-1 answered SIP/2003-08589910
-- Hungup 'Zap/1-1'


What to do next??

 

[Bart]

Looks like it's dialing out - what happened? Do you really need 0 prefix? Called g0/0xxxxxxxx

Where are you located?

Bart

 

[v0ip-Pete]

Based in UK, dialing UK mobile 07xxx xxxx
which is classed as national number. Full international code would be 00447xxx xxx

 

[cosmicwombat]

In the first trunk..

try a 1. Then for the second trunk use a 2.

Make sure to create an outbound route that is appropriate.

 

[v0ip-Pete]

Nope, still the same, prefixed with a 1 for line1.

 

[cosmicwombat]

what do you mean prefixed?

in the trunk setting...you should have tried a 1 for channel.

 

[v0ip-Pete]

Not sure what you mean, In my trunk setting I have
Zap Identifier (trunk name): have set this to g0 and g1.

Makes no difference

 

[bgriff]

Did you try the trunk identifier without the g in front of the 1 and 2? My zap trunks just have a 3 or 4 for the respective channel. I think that is what Robert was saying also.

 

[cosmicwombat]

That is exactly what I was saying.

That is exactly what I was saying.

 

[v0ip-Pete]

Thanks guys,

tried that, same result, no ringing and console ends with
Zap1-1 answered.

I think I have the wrong modules in the card - I have green FXS, I think I need the red FXO ones to connect to PSTN??

 

[Alex728]

yep you need the red FXO ones to connect

The config files are confusing as they deal with the signalling protocol to the hardware, not the hardware itself

they show FXS (phone) signalling to FXO (a trunk module)

and FXO (phone line) signalling to FXS (an extension module)

if you think about it this way, with your normal phone you signal with this telephone set (by picking up the receiver, dialling or answering a ringing telephone) to the FXO line

if you have a Zap extension, Asterisk needs to be the Telephone Exchange (FXO) and provide dialtone, ringing, -50 volt battery etc so you can plug in a telephone

 

[v0ip-Pete]

Thanks Alex,

So it's definately the red FXO modules I need to connect to a PSTN trunk for dialing and receiving calls in an out of Asterisk?


Green FXS are for normal phone extensions, which I dont need. (and come to think of it why would anyone want to connect a plain old analogue phone to the system when the IP phones have much more features??)

 

[Alex728]

Thanks Alex,

So it's definately the red FXO modules I need to connect to a PSTN trunk for dialing and receiving calls in an out of Asterisk?

indeed. Incidentally the atcom ones are red too (just checked a photo of mine I took last year)...

Green FXS are for normal phone extensions, which I dont need. (and come to think of it why would anyone want to connect a plain old analogue phone to the system when the IP phones have much more features??)

I see what you mean but there a still fair few reasons for doing this.

you may have a phone in a lobby or for public use or in a industrial workshop where an IP phone may not be robust enough or at risk of being nicked.

That said i prefer to use Linksys ATA's for this purpose as they are more easily configureable and can be located remotely from the server (wherever there is a network cable and electric supply).

Other end users (particularly those in non techie jobs) find an IP phone or even a key system phone to be complicated.

For the office system I gave the users a choice of Linksys SPA942s and analogue phones and the users in this office wanted SPA942s but the others wanted analogue (cordless DECT) telephone sets!

This is because they often walk to a nearby filing cabinet (its a small office) and can remain on the phone to the customer, plus some thought "all those buttons? it is confusing, we just want a telephone."

To "insist" they had desk phones would reduce customer service as they would be put on hold for longer..

(I know you can now get DECT IP telephones but they are way expensive here...)