R.I.P. No Dialtone on fxs extension

[tiggerpaws]

I'm trying to set up an analogue extension
but I get no dialtone I have a openvox a400p
card.

Dahdi seems to be running, I queried linux and the
asterisk cli for it and seems recognised.

I will post as mjuch info as Ii remmeber if I
leave any out Please ask and I will give it.

system info:
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = OFFLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PIAF Installed Version = 2.0.6.4 under *HARDWARE* │
│ FreePBX Version = 2.10.0.2 │
│ Running Asterisk Version = 1.8.23.0 │
│ Asterisk Source Version = 1.8.23.0 │
│ Dahdi Source Version = 2.7.0 │
│ Libpri Source Version = 1.4.12 │
│ IP Address = 192.168.0.78 on eth0 │
│ Operating System = CentOS release 6.4 (Final) │
│ Kernel Version = 2.6.32-358.6.2.el6.i686 - 32 Bit



DAHDI Tools Version - 2.7.0

DAHDI Version: 2.7.0
Echo Canceller(s): MG2
Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)

3 channels to configure.

Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
Setting echocan for channel 3 to mg2


Connected to Asterisk 1.8.23.0 currently running on pbx (pid = 2623)
Verbosity is at least 3
pbx*CLI> dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)
pbx*CLI> dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)
pbx*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
pbx*CLI>

im tring to set up an extension on channel 2
i typed a '2' where it says channel in the
freepbx gui I tried both a dahdi and a zaptel
kind of extension, what do I do
and what mmore info do you need?

 

[MGD4me]

Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)

It would appear that you should be connecting the analogue phone to Channel 1, not Channel 2. Channel 2 (FXO) would be for interfacing to a POTS (telephone) line.

 

[MGD4me]

I see that you also cross-posted your question in the Forum's Trunks section... tsk-tsk...

 

[tiggerpaws]

It would appear that you should be connecting the analogue phone to Channel 1, not Channel 2. Channel 2 (FXO) would be for interfacing to a POTS (telephone) line.

So I need to put the fxo on channel 2 instead?
I have the red fxo boardlet on slot 1 and then a green fxs boardlet on slot 2 and
another on slot 3, so this is not correct?

If it would help I will switch the fxo board to slot 2 and the green fxs one
back to slot 1, so do I leave the fxs green one that is in slot 3 where it is?

 

[tiggerpaws]

Well I switched the boardlets everywhich way but I don't know and it still don't work.

It is detecting the board, but under asterisk cli it says:
Connected to Asterisk 1.8.23.0 currently running on pbx (pid = 1569)
Verbosity is at least 3
pbx*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service

I don't understand what is wrong.
I set up the extension in freepbx
as a zap (dahdi compatible)
then tried it with dahdi, and still no
dial tone.

 

[tiggerpaws]

Ok, downloaded the dahdi module updates on the
module admin.

 

[tiggerpaws]

Its a long story over the past day or 2, but I am giving up on pbxing all together.
many thanks to all who tries to help.
i am discontinuing all my phone lines and possibly shelving
or selling our equipment, I cannot do this anymore,
i am retiring and I feel too alone in this kind of
thing to go on.

thanks to all who helped and for letting me borrow
your software and ideas.

 

[TwigsUSAN]

Wait a minute. How long have you been trying to get this up and running? Why do you need an FXS port?

 

[tiggerpaws]

An FXS port is required to connect an analogue phone,
which now works, but now my other (old ) pbxsystem
is @#$@#$ up and won't work at all.

Our system is totally down and I have no pooping clue
how to fix it again.

Our vitelity trunk is just dead. no in, no out, don't work
and i don't know how to fix it.

I can use the google voice lines, strangely enuf,
but that is all that is configged on piaf box.

 

[TwigsUSAN]

Are your Vitelity trunks showing up registered? Could you replace the analog phone with a voip phone?

 

[tiggerpaws]

I cannot tell if the vite trucnks are regustered because
I use the stati ip option.
I don't have a voip phone.

 

[TwigsUSAN]

Try using a soft phone like X-lite. You will be able to test out the Vitelity lines that way.