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× Questions about A400/800/1200 Analog Interface Card

Analog phone on FXS port - No ringing -> straight to voicemail

14 years 2 months ago #4954 by mwilson75
I installed an OpenVox a800p for some testing because I've never used any Dahdi hardware before.
Outbound calls to extensions and PSTN work fine. Inbound calls to the dahdi ext go straight to voice mail.
The system is:
Ubuntu 9.10
Asterisk 1.4.2.8
Dahdi Linux and Tools 2.2.1
Port 1 on the a800p is FXO
Port 2 on the a800p is FXS
Again,...Outbound calls to the PSTN and other extensions work fine from the analog phone I have on port 2. But I cannot call the ext inbound from another extension or the PSTN.
The analog phone is set up as a ZAP extension with:
channel 2
context from-internal
signaling fxo_ks
dial ZAP/1
chan_Dahdi.conf looks like this:
[channels]
language=en
; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf
; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
signalling=fxo_ks
context=from-internal
channel => 2
signalling=fxs_ks
context=from-internal
channel => 1
/etc/dahdi/system.conf looks like this:
fxsks=1
fxoks=2
loadzone = us
defaultzone=us
echocanceller=kb1,1
echocanceller=kb1,2
The only clue I have is this piece in the Asterisk log:
[Mar 18 00:35:32] WARNING[1920] channel.c: No channel type registered for 'ZAP'
[Mar 18 00:35:32] WARNING[1920] app_dial.c: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented)
WHAT AM I DOING WRONG??
14 years 2 months ago #4955 by xin.liu
hi, mwilson75
I think your extensions.conf has some problem,the type of channel is dahdi,not zap,please check it.

14 years 2 months ago #4956 by Denins.Den
I think you are using dahdi, so when your dialplan should like this:

Dial(dahdi/1/xxxx)

Please notice " not zap "
14 years 2 months ago #4957 by mwilson75
I changed the "Dial" setting in FreePBX extension to dahdi/2 and now it rings and accepts calls.

But, ...caller ID shows unknown when I call from Analog phone to SIP extensions. Also in FreePBX reports clid and source are blank.

Any ideas on those 2 problems??
14 years 2 months ago #4958 by xin.liu
Hi:
1) set loadzone=your country, defaultzone=your country in /etc/dahdi/system.conf
2) set country=your country in /etc/asterisk/indications.conf
3) load driver with your country code:modprobe opvxa1200 opermode=YOUR COUNTRY
4) set this in chan_dahdi.conf
cidstart=ring
cidsignaling=dtmf
usecallerid=yes
if it still has issue, you can check whether your PSTN line is provided with caller ID function.

14 years 2 months ago #4959 by Denins.Den
you should check the extensions config in web gui
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