TDM410P: cannot use one POTS line

Hello

Asterisk 1.4.21.2 (AstLinux 0.6.1) + TDM404EF (=TDM410P + 4 FXO + hardware echo cancellation). Currently using only channels 1 and 2, both plugged into separate UK POTS lines. Using wctdm24xxp driver with opermode=uk.

zaptel.conf:

[code]fxsks=1
loadzone=uk
defaultzone=uk

fxsks=2
loadzone=uk
defaultzone=uk[/code]

zapata.conf:

[code][channels]

usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes

context=incoming
signalling=fxs_ks
group=1
channel => 1

context=incoming
signalling=fxs_ks
group=1
channel => 2[/code]

extensions.conf:

[code][work]
exten => _0.,1,Dial(Zap/g1/${EXTEN})

[incoming]
exten => s,1,Dial(SIP/UserA,20,t)[/code]

sip.conf:

[UserA] type=friend host=dynamic context=nnpa username=UserA secret=password disallow=all allow=gsm allow=ulaw mailbox=0234 callerid=Test user <0121 111222>

Only one line works, the other failing for both incoming and outgoing. I can rule out the card or the modules, because if I swap which module is connected to which POTS socket, the channels which work from Asterisk’s point of view swap over.

Am I doing something obviously wrong? Do I have to pass any other parameters alongside opermode? As is apparent, I’m a beginner.

Thanks

Tom

Update:

Very odd: now the line which was not working is working, and the line which was working does not. What on earth is going on?

I have the TDM410P plugged in via sockets in my office which are patched down through the building to the telco master sockets. Could this be causing a problem?

Any advice greatly welcome

Tom

I moved the Asterisk box down to the master POTS sockets, and plugged the TDM410P directly in. No difference. The first outgoing lineworks, but the second fails, with three of this error:

WARNING[1737]: translate.c:175 framein: no samples for gsmtolin

Incoming calls work on the first line. On the second line, the external calling party hears a ringing tone, but there is no activity on the Asterisk console.

Can anyone help me?

Thanks

Tom

Hi

what does zttool say and what does zap show channels report ?

Ian
www.cyber-cottage.co.uk

zttool:

[code]Alarms Span
OK Wildcard TDM410P Board 1

Current Alarms: No alarms.
Sync Source: Internally clocked
IRQ Misses: 3
Bipolar Viol: 0
Tx/Rx Levels: 0/ 0
Total/Conf/Act: 4/ 4/ 1

  1234

TxA ----
TxB ----
TxC ----
TxD ----

RxA ----
RxB ----
RxC ----
RxD ----[/code]

zap show channels:

Chan Extension Context Language MOH Interpret pseudo incoming default 1 incoming default 2 incoming default 3 incoming default 4 incoming default

Any use?

Thanks

T

Ok

change zaptel and zapata to

fxsks=1-4 loadzone=uk defaultzone=uk

zapata.conf:

[code][channels]

usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes

context=incoming
signalling=fxs_ks
group=1
channel => 1-4[/code]

then reload zaptel and asterisk

then do zap show channel 1 etc

Channel: 1
File Descriptor: 13
Span: 1
Extension:
Dialing: no
Context: incoming
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Offhook

Channel: 2
File Descriptor: 14
Span: 1
Extension:
Dialing: no
Context: incoming
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

Channel: 3
File Descriptor: 15
Span: 1
Extension:
Dialing: no
Context: incoming
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

Channel: 4
File Descriptor: 16
Span: 1
Extension:
Dialing: no
Context: incoming
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

Channels 1 and 2 are connected to the PSTN. Channels 3 and 4 are not connected to anything.

Thanks

T

Ok so they all show up

Could you set verbose and debug to 99 then fire a call into each line. and post any output for each

Also change you dialplan so that it uses Zap/1 and then Zap/2 so it is forced to use line 1 nad 2 and not groups.

Ian

extensions.conf now has:

exten => 111,1,Dial(Zap/1/www01434600000,10)
exten => 222,1,Dial(Zap/2/www01434600000,10)

Call using first PSTN line:

-- Executing [111@work:1] Dial("SIP/UserA-0820c958", "Zap/1/www01434600000|10") in new stack
-- Called 1/www01434600000

Really destroying SIP dialog ‘6b1770cc0e4bd92950cee1d9154edc1b@10.8.243.4’ Method: NOTIFY
Really destroying SIP dialog ‘153bdea734da627a3ab0894c245d9f95@10.8.243.4’ Method: NOTIFY
– Zap/1-1 answered SIP/UserA-0820c958

There is a brief pause, and then the recipient PSTN external number starts to ring correctly.

Call using second PSTN line:

-- Executing [222@work:1] Dial("SIP/UserA-082307e8", "Zap/2/www01434600000|10") in new stack
-- Called 2/www01434600000
-- Zap/2-1 answered SIP/UserA-082307e8

Nothing then happens. Silence, and the handset doesn’t ring.

Thanks for all the help - it’s got me stumped.

T

PS I should say that sip.conf and extensions.conf are copied over from our existing VoIP-only set-up, with many users and extensions, so some of the extra SIP messages in there could well be to do with that.

Hi Have you got a test Butt that you can listen to the line with ?
also try swapping the lines over , and repeating the tests with trunk 3 and 4 , It could be a faulty module or faulty line. HArd to say as it seems to be doing everting correct, What shows up when you call each line.

You are down no to normal fault finding ie swapping modules over.

Ian

I have purchased a TDM410P with single FXO to replace an X100P which has worked for over 2 years and releases of Asterisk and Zaptel upto 1.4.21.1.

Found a similar problem with dialing out and not picking up incoming calls.

All diagnostics show a card and channel.

I need to get this card working and UK Callerid without the usehist patches for Zaptel ( which I apply manually for the X100P), so that I can move from 1.4.21.1 to 1.6. If a cannot get this card to work, I will not be able to move forward with new releases of Asterisk and Dahdi. Tried to apply the old zaptel usehist patches to Dadhi without any success.

Can anyone confirm also, whether this card with show CallerID in the UK

thanks

Hi
The usehist patch was for x100p cards , the TDM400 should support UK cli OK, what have you set in the conf file?

on a side not, Its not a good idea to tack an unconnected question onto a thread… you stand a high change of it being missed.

Ian

thanks for the aside reply !!

My main issue is that the TDM410p card/asterisk does not dial or receive calls.

The configuration for the X100P card has always worked ok, so was assuming that the TDM410P card would work with the same zaptel/zapata configurations.

Asterisk sees ZAPTEL - the CLI shows the SIP extension connection, but does not appear to dial out.

I have included all the zaptel modules, as I have seen different modules being specified in forum threads as to which module is required for this card.

my configs are :

ZAPATA.CONF
[channels]
language=en
context=BTInbound
signalling=fxs_ks
busydetect=no
busycount=4
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callprogress=no
echocancel=yes
echotraining=yes
rxgain=4.5
txgain=4.5
immediate=no
faxdetect=both
faxdetect=incoming
faxdetect=outgoing
mohsuggest=default
mohinterpret=default
callerid=asreceived
hidecallerid=no
usecallerid=yes
cidsignalling=v23
cidstart=usehist
callwaitingcallerid=yes
callgroup=1
pickupgroup=1
group=1
channel => 1

ZAPTEL.CONF
loadzone = uk
defaultzone=uk
fxsks=1

I cannot give any other diagnostics, at present, as my system is currently running live with the X100P card. If diagnostics would help, I would have to change the card and restart .

With the TDM410P is place, the card lights are present, and have tried numberous different configurations with KEWLSTART and LOOPSTART, all showing errors, proving that my current conf’s appears to be ok. Have turned on Verbose and full debug, not no obvious errors are displayed.

At a loss, to know what to do next.

I also have a TDM01B card, which I bought 12 months ago, and thought it may be faulty etc, to bought this newer card instead, but both appear to be having similar issues.

Sadly not, and I can’t really call out a BT engineer, since a handset plugged directly into the master socket works fine, so I guess he would find no fault, and charge us for the privilege.

Swapped the lines over. Using the same dialplan, 111 now acts as 222 did - failing - and 222 acts as 111 did - ringing.

I’ll do this next.

This is what is intriguing - it can’t be a faulty module, since each module works with one of the lines. It also can’t be the line, since an analogue handset works with the other line.

Calling the line which works rings the SIP phone correctly. Calling the other line, I get a ringing tone in the external handset, but see no activity on the Asterisk console.

I’ll try this, too.

Many thanks for all the help. I really appreciate it.

Tom

Hi If this is a new card then call Digium support, Or if the supplier gives support call them, As a Supplier myself I do assist my customers to a point and in 99% of cases solve the issue, so yours should be no different but if you have no luck get in touch with digium, ( you will need to let them login though)

Ian
www.cyber-cottage.co.uk

Hi Ian

I already have a ticket open with them via email, but they are v slow to reply. I’ll see what that yields in time.

Thanks for all the suggestions

Tom

I’ve spoken to my supplier too !

But no help so far.

rather than piggybacking on this thread I will raise a new thread

there must be someone somewhere who knows why these cards are not working in the UK

thanks

To Both of the Above.

Ring them up, Obvoiusly has to be done in the evening, But as to UK working, They do work fine in the UK.

@iasgoscouk your main problem is using incompatible entries in the zapata.conf and patching when not needed, best be is the recompile zaptel and asterisk with no patches.

Ian

thanks for the reply - would be interesting to know what the incompatable entries are.

I am very dissappointed with digium and asterisk. there is little or no documentation to cover this type of issue and the move/rename from Zaptel to dahdi has not been that invisible. I have worked on a number of asterisk installations in the UK with this configuration, and I am now left with something which cannot be easily answered.

Also the downside to the forums - everyone asks for help - but no-one ever publishes the fix. !!!

I am currently not in an immediate position to prepare a separate installation without patches. This is for my personal implementation and not for any others.

a very disappointing outcome !

Hi

cidstart=usehist This was for Marc’s patch for the x100p

you might want to look at forums.digium.com/viewtopic.php? … b401#27484

A key problem I identied in the UK was the different exchange types, Things may work fine on a DMS100 but on a TXE they fail and on a Nokia it was intermittent.

If I had time I may look into this again I have a card in a lab system so wouldnt be too much trouble. Its just time realy

Ian