So I've setup an asterisk 1.6 box in our office with the hope of binning our legacy PBX. Managed to get everything running except busy detect just won't work.
I got some advise here on the settings that 'should' work
Hoping that some asterisk guru can assist.
Here's the relevant config files:
/etc/asterisk/chan_dahdi.conf:
/etc/dahdi/system.conf :
/etc/asterisk/indications.conf
I've tried these things already without any change:
* fxs_ks signalling
* the siemens dial/ring/callwaiting instead of alcatel
* busypattern=2500,500 in chan_dahdi.conf
* installed asterisk 1.4 & zaptel from debian repo with the same results
If the caller hangs up when there is silence it seems to work properly, but if they hangup while in a voice prompt or holding music it never clears the channel.
I'm about to give up on getting these analogue lines usable on asterisk, any help would be appreciated.
Oh, we're in Cape Town, if I listen in to what happens after the call disconnects there is a busy signal, about 1/2 sec repeated.
I got some advise here on the settings that 'should' work
Hoping that some asterisk guru can assist.
Here's the relevant config files:
/etc/asterisk/chan_dahdi.conf:
Code:
[trunkgroups]
[channels]
language=en
context=incoming
echocancel=no
echocancelwhenbridged=no
busydetect=yes
busycount=4
busypattern=500,500
callprogress=no
signalling=fxs_ls
callerid=asreceived
group=0
context=from-pstn
channel => 1-4
/etc/dahdi/system.conf :
Code:
fxsls=1
echocanceller=mg2,1
fxsls=2
echocanceller=mg2,2
fxsls=3
echocanceller=mg2,3
fxsls=4
echocanceller=mg2,4
# Global data
loadzone = za
defaultzone = za
/etc/asterisk/indications.conf
Code:
[general]
country=za ; default location
...
[za]
description = South Africa
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
; (definitions for other countries can also be found there)
; Note, though, that South Africa uses two switch types in their network --
; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer
ringcadence = 400,200,400,2000
; dial/ring/callwaiting for the Siemens switches:
;dial = 400*33
;ring = 400*33/400,0/200,400*33/400,0/2000
;callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
; dial/ring/callwaiting for the Alcatel switches:
dial = 383+417
ring = 383+417/400,0/200,383+417/400,0/2000
callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
congestion = 400/250,0/250
busy = 400/500,0/500
dialrecall = 350+440
; XXX Not sure about the RECORDTONE
record = 1400/500,0/10000
info = 950/330,1400/330,1800/330,0/330
stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33
...
I've tried these things already without any change:
* fxs_ks signalling
* the siemens dial/ring/callwaiting instead of alcatel
* busypattern=2500,500 in chan_dahdi.conf
* installed asterisk 1.4 & zaptel from debian repo with the same results
If the caller hangs up when there is silence it seems to work properly, but if they hangup while in a voice prompt or holding music it never clears the channel.
I'm about to give up on getting these analogue lines usable on asterisk, any help would be appreciated.
Oh, we're in Cape Town, if I listen in to what happens after the call disconnects there is a busy signal, about 1/2 sec repeated.


