Asterisk box does not answer incoming PSTN call

Asterisk Asterisk Support
1

Hello everyone.
I am looking for assistance.

Here is what I have.
Asterisk 1.4.8 loaded and running.
I also requested sample conf files at the time of the install.

Using wctdm24xxp with 1 FXO module installed.
I have a requirement of needing only 4 analog lines coming
from my local telco.

I have zero FXS modules installed.

I will not be using any analog phones. I am using grandstream IP phones
for all extensions. Hence, no FXS, module(s)

Zaptel.conf

fxsks=1-4
loadzone=us
defaultzone=us

ztcfg -vv yields the following:

Channel map:
Channel 01 : fxs Kewlstart (default) (slaves: 01)
Channel 02 : fxs Kewlstart (default) (slaves: 02)
Channel 03 : fxs Kewlstart (default) (slaves: 03)
Channel 04 : fxs Kewlstart (default) (slaves: 04)

4 channels configured

Zapata.conf

usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
context=incoming
signaling=fxs_ks
channel= 1-4

Please note: I have not identified any FXS devices. I am not using analog phones in this system, only IP phones.

Extensions.conf

[incoming]
exten=>s,1,Answer()
exten=>s,2,Hangup()

I have the first tip/ring pair coming directly from my local telephone tied to pin 1 and 26 (channel 1) OF MY wctdm24xxp card.

When I call the local telephone company number tied to my Asterisk box all I want to happen at this time is that the Asterisk box answer the call and hang up. Asterisk never answers the call. I have opened the connection at the asterisk box and can short the pair back towards the local telco and answer the call.

I am not using Asterisk now at this point. I have had problems with Asterisk now Beta 5 so I thougth I’d try the command line version.

With command line interface I am not sure where to enter service providers and incoming calling rules. I am thinking that without this information the call will not be answered. Someone told me that asterisk now is not reliable. This is why I am using command line 1.4.8 version of Asterisk. I am not sure the teller of this tale was accurate. Is Asterisk Now beta 6 ok to use? Any help with this lack of Asterisk not answering an incoming call would be most appreciated. I’ll build the extensions next.
Thanks,
davemo

2

Hi

What does zap show channels show you?
What does zttool say about the card
what do you see on the CLI when a call comes in?

and finaly are you sure that the ports configured are the ones that the module is installed in

Ian

3

Hello Ian,
Thanks for the quick response.

zttool tells me:

Alarms Span
OK wildcard tdm2400P rev e/f board 1

The FXO module is plugged into slot 1.

I see no messages of any kind at the CLI.
I have input the following:
set debug 10
set verbose 10

However, I am not sure the commands were honored.
I am up on Linux Fedora 7.
Possibly I should be Red hat??
I need to verify zap show channels. Not 100 % sure on this one.
Thanks again
davemo

4

So far so good

what does zttool show when you select the tdm2400 ?

as to “zap show channels” just enter it at the * cli and it will respond back with what the channels are configured as.

Ian

5

Thanks Ian,
I will return here with the zap show channels information as soon as possible.
I will also provide additional stats on tdm2400 VIA zttool.
Again,
Thanks for your help…
Davemo

6

Hello Ian

Zap show Channel follows:

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

ocalhostCLI> zap show channel 1
Channel: 1CLI>
File Descriptor: 16
Span: 1
Extension: LI>
Dialing: no
Context: incoming
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0tCLI>
Owner:
Real:
Callwait:
Threeway:
Confno: -1CLI>
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, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

Any assistance would be appreciated…
davemo

7

Also channels 2,3 and 4 look the same…
davemo

8

Hi

Set the dialplan to

[incoming]
exten=>s,1,Answer()
exten=>s,n,Playback(tt-weasels)
exten=>s,n,Hangup()

So at least somthing has to happen.

Then connect to the * cli, set debug to 99 and verbose to 99 and fire a call in.

Ian

9

Hello Ian,
Finally getting back to the problem. I added the lines of code to extensions.conf and got the required answer. I also sent an extensions reload command from the CLI just prior to the box answering the call. I found the note on the reload command in the verbose within extensions.conf file.
Thanks for your assistance with this. It made my job much easier.
FYI… I am adding 16 extensions to this system. All are Grandstream IP phones. I have added 2 of the 16 and they came up after building the SIP.conf files. It looks like command reload chan_sip.so needs to executed after any changes to the sip.conf file. It that the way you see it?? Also it appears that command reload chan_zap.so needs to be executed after any changes to zapata.conf.zapscan. I believe this is an excellent system after tearing out what is left of my original hair. Seriously though… Asterisk will serve our customer well.
Thanks again, If I see you in the field sometime. I’ll buy coffee.
davemo