Hello all:
I have a new install with Asterisk V 1.4.13, zaptel V 1.4.6 and libpri V 1.4.2 on a CentOS V 5.0 Linux and the following hardware Linksys SPA3102, Linksys PAP2T-NA, X100P.com X100P FXO PCI cards x 2, and one X100P.com S100-FX.
I have the FXS side working fine. I can attach an analog phone to the S100-FX, the SPA3102 or the first port of the PAP2T and call between them or to SIP or iax2 soft phones.
But I cannot configure the FXO side of things. I don’t care if I use the SPA3102 or one of the X100Ps at the moment as I only have one PSTN line, but that will change.
I would greatly appreciate it if somebody would do a sanity check on my config files or suggest some diagnostic steps.
If you have a very basic set of files that works with any combination of this equipment, I would like to see them.
Thank you: John Turnbull
Below are some diagnostic messages and my conf files:
I can see a Wildcard FXO: Wildcard X100P in dmesg
zttools show an: OK Wildcard X100P Board 1
lspci finds: 00:0f.0 Communication controller: Motorola Wildcard X100P
lsmod show the following relevant things:
wctdm 37184 0
wctdm24xxp 103616 0
wcte11xp 26656 0
wct1xxp 17184 0
wcte12xp 37696 0
wct4xxp 298048 0
zaptel 187428 12 zttranscode,wcusb,wctdm,wctdm24xxp,
wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2,wcfxo
but
ztcfg -vv shows:
Zaptel Version: 1.4.6
Echo Canceller: MG2
Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels to configure.
So that might be the problem with the Z100P. How do I fix this?
===
My /etc/zaptel.conf:
fxsks=1
loadzone=us
defaultzone=us
===
My /etc/asterisk/zapata.conf:
[trunkgroups]
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
context=incoming
signaling=fxs_kse
group=1
channel => 1
===
My /etc/asterisk/sip.conf:
[general]
[1000 ; X-Lite phone on D610
type=friend
context=phones
host=dynamic
[1001] ; ZoIPer SIP on D610
type=friend
context=phones
host=dynamic
[1005] ; Linksys SPA3102 phone type=friend
; this phone can call in and out
context=phones
host=dynamic
[1006] ; Linksys PAP2T phone #1 Exten. 1006
type=friend
context=phones
host=dynamic
[1007] ; linksys Pap2T phone #2 Exten. 1007
; This is bust, and I don’t know why
type=friend
context=phones
host=dynamic
===
My /etc/asterisk/iax.conf
[general]
autokill=yes
[1002] ; ZoIPer iax2 phone on D610
type=friend
username=1002
host=dynamic
context=phones
[1010] ; The extension for the S100-FX phone
type=friend
username=1010
host=dynamic
context=phones
; peercontext=default ; from the X100P.com website. No idea of function
qualify=no
disallow=all
allow=all
===
My /etc/asterisk/extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait()
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[incoming]
exten => s,1,Wait()
exten => s,n,Answer()
exten => s,n,Playback(T-to-leave-msg)
exten => s,n,Verbose(1|Incoming call from PSTN)
exten => s,n,Hangup()
[internal]
exten => s,1,Answer()
exten => s,n,Wait()
exten => s,n,Playback(why-no-answer-mystery)
exten => 500,1,Verbose(1|Have 500 say something)
exten => 500,n,Playback(tt-allbusy)
exten => 500,n,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
exten => 1000,1,Verbose(1|Somebody called 1000, X-Lite SIP on D610)
exten => 1000,n,Playback(silence/2) ; 2 sec. silence to sync phones & sound
exten => 1000,n,Dial(SIP/1000,10)
exten => 1000,n,Hangup()
exten => 1001,1,Verbose(1|Someone called 1001, ZoIPer SIP on D610)
exten => 1001,n,Playback(silence/2)
exten => 1001,n,Dial(SIP/1001,10)
exten => 1001,n,Hangup()
exten => 1002,1,Verbose(1|somebody called 1002, ZoIPer iax2 on D610)
exten => 1002,n,Playback(silence/2)
exten => 1002,n,Dial(IAX2/1001,10)
exten => 1002,n,Hangup()
; extension 1003 does not work
exten => 1003,1,Verbose(1|Forced call to weather on PSTN with X100P)
;exten => 1003,n,Wait(1.5)
exten => 1003,n,Playback(silence/2)
exten => 1003,n,Playback(moo2)
exten => 1003,n,Dial(Zap/1/6139983439)
exten => 1003,n,Playback(vm-nobodyavail)
exten => 1003,n,Hangup()
exten => 1004,1,Verbose(1|Ring a phone connected to IAX2 S100-FX) ;
exten => 1004,n,Playback(silence/2)
exten => 1004,n,Dial(IAX2/1010,10)
exten => 1004,n,Wait(1.5)
exten => 1004,n,Playback(enter-ext-of-person)
exten => 1004,n,Wait(1.5) ;
exten => 1004,n,Hangup()
exten => 1005,1,Verbose(1|Ring a phone connected to SIP SPA3102)
exten => 1005,n,Playback(silence/2)
exten => 1005,n,Dial(SIP/1005,10)
exten => 1005,n,Wait(1.5)
exten => 1005,n,Playback(weather)
exten => 1005,n,Hangup()
exten => 1006,1,Verbose(1|somebody called 1006, PAP2T SIP phone #1)
exten => 1006,n,Playback(silence/2)
exten => 1006,n,Dial(SIP/1006,10)
exten => 1006,n,Hangup()
; This does not work either, but not high importance
exten => 1007,1,Verbose(1|somebody called 1007, PAP2T SIP phone #2)
exten => 1007,n,Playback(silence/2)
exten => 1007,n,Dail(SIP/1007,10)
exten => 1007,n,Hangup()
exten => 1010,1,Verbose(1|Sombody called the S100-FX on 1010)
exten => 1010,n,Playback(touchtone2)
exten => 1010,n,Hangup() ;
[phones]
include => internal
And I have a bad memory…
Alot more then I know, ive never had to go through all that. I dont know enough to help out anymore. I would suggest returning them if you can.