I thought I would post here before heading over to the mailing list. I am sorry if this is actually a channel bank question and not an asterisk configuration question.
I have an asterisk 1.2 system with a te410p, an incoming t1 line in span 1, and a zhone channel bank in span 2. The zhone sucks at end of call disconnect, and I have had to use these thingies so that my obscure legacy dial in modem system actually hangs up.
Anyway, I was digging around our old phone system and came across an adit 600 with some FXS 8A cards. So I got to thinking maybe this would make a better channel bank than the zhone we are currently using.
Alright here we go
[code]> vi /etc/zaptel.conf
span=1,1,0,esf,b8zs,yellow
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs,yellow
fxols=25-48
loadzone = us
defaultzone=us[/code]
[code]> vi /etc/asterisk/zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=no
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6
signalling = pri_cpe
group = 1
channel => 1-23
usecallerid=yes
signalling = fxo_ls
group = 2
channel => 25-48[/code]
Reboot asterisk server
Alright lets set up the adit
[code] > minicom
restore defaults
reset
set local off
Set a:1 down
Set a:2 down
Disconnect a
Set a:1:1-24 type voice
Set a:1:1-24 signal ls
Set 1:1-8 signal ls
Set 2:1-8 signal ls
Set 3:1-8 signal ls
Connect a:1:1-8 1:1-8
Connect a:1:9-16 2:1-8
Connect a:1:17-24 3:1-8
Set clock1 internal
Set a:1 up[/code]
That wasn’t that bad now lets see.
[code] > show a:1
SLOT A:
Settings for DS1 1:
Circuit ID: CAC DS1# A:1
Up/Down: UP
Framing: ESF
Line Coding: B8ZS
Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB)
Loop Code Detection: CSU
Loopback: OFF
FDL Type: None
AIS Forwarding: Disabled
AIS List: all
Performance Threshold Settings 15 min. 1 day
--------- ----------
unavailable seconds: 0 0
severely errored seconds: 0 0
errored seconds: 0 0
sev. errored frame seconds: 0 0
line errored seconds: 0 0
controlled slip seconds: 0 0
bursty errored seconds: 0 0
degraded minutes: 0 0
total linecode violations: 0 0
total pathcode violations: 0 0
show a:1:1
SLOT A:
Settings for DS1 1: Channel 1:
Type: VOICE
Signaling: LS
Interface Side: DROP
show 1
SLOT 1:
Settings for FXS: Channel 1:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 2:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 3:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 4:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 5:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 6:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 7:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
SLOT 1:
Settings for FXS: Channel 8:
Signaling: LS
RxGain: -6 dB
TxGain: -3 dB
Impedance: 900 Ohms + 2.16uF
status a:1
SLOT A:
Status for DS1 1:
Receive: Traffic
Transmit: Traffic
Loopback: OFF
status a:1:1
DS0 Rx AB Tx AB Signal T1 TP
a:1:1 01 01 LS Traffic N
status 1
FXS Rx AB Tx AB Signal=>T1 Sig T1 TP
1:1 01 01 LS => LS Traffic N
1:2 01 01 LS => LS Traffic N
1:3 01 01 LS => LS Traffic N
1:4 01 01 LS => LS Traffic N
1:5 01 01 LS => LS Traffic N
1:6 01 01 LS => LS Traffic N
1:7 01 01 LS => LS Traffic N
1:8 01 01 LS => LS Traffic N[/code]
That looks right?
At this point I punched in a line to connect to an analogue phone, and horrays o/ I have a dial tone and can dial the sip lines.
Alright lets test calling the analogue lines
[code]> vi /etc/asterisk/extensions.conf
exten => 725,1,Dial(Zap/25)
:w :q
asterisk -rc “extensions reload”
[/code]
Then dialing 725 results in … NO Ring! The led on the fxs card is blinking correctly. But it is not ringing, however if I pick the analog line up the call connects.
Alright lets take a listen to what the phone is getting.
The adit sounds like this adit.mp3
One click, then two clicks, followed by the callerid, followed by a repeating One click, then two clicks
The ring should sound something like zhone.mp3
Ring pulse, followed by the callerid, followed by more ringing.
Yeah something is wrong, Let us fire up ztmonitor
aditdump.txt
zhonedump.txt for comparison
Something is wrong indeed.
Any advice? I’m at a loss, I think I am missing something glaringly obvious, or I have a hardware failure with the ring generation on the adit.
I can try to it with mgcp as well, but I need a nap before diving into the 668 page adit manual.
edit:
It was a hardware problem after all, the pins connecting the power supply to the backplane were all bent. Tweasers save the day.