Salve.
Ho configurato un server asterisk a puro scopo di test,
ho messo 2 telefoni voip,e in casa mia ci sono 3 prese analogiche, tutte occupate, una dal centralino.
Le altre da telefoni analogici.
Al centralino sono collegati i 2 telefoni voip.
Il server funziona bene,ma mi da questo problema.
Se rispondo con un telefono voip gli analogici smettono di squillare(bene)
ma rispondo con uno dei 2 analogici, i telefoni ip continuano a squillare?
Come è possibile far riconoscere a asterisk che è stato risposto a un altro apparecchio?
Ecco i miei file di conf

chan_dahdi.cofn

codice:
[trunkgroups]

[channels]
context=from-pstn
language=it
signalling=fxs_ks
rxwink=300              ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
faxdetect=incoming
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
;Addesd
dialtone_detect = yes ; Watch for dialtone for 10 seconds after answer
dialtone_detect = always ; Watch for dialtone for the whole call


;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=3


immediate=no
#include /etc/asterisk/dahdi-channels.conf
sip.conf
codice:
[general]
context=mycontext
bindaddr=192.168.0.7
disallow=all   ; First disallow all codecs
allow=ilbc
allow=alaw
allow=ulaw
allow=gsm
allow=g722
allow=speex
allow=h261
language=it
srvlookup=yes
videosupport=yes
tos=lowdelay
[1001]
type=friend
username=giuseppe
secret=secret
callerid=giuseppe
host=dynamic
canreinvite=no
qualify=yes
[1002]
type=friend
username=giuseppe2
secret=secret
callerid=giuseppe2
host=dynamic
canreinvite=no
qualify=yes
[1003]
type=friend
username=giuseppe3
secret=secret
callerid=giuseppe3
host=dynamic
canreinvite=no
qualify=yes
[1004]
type=friend
username=giuseppe4
secret=secret
callerid=giuseppe4
host=dynamic
canreinvite=no
qualify=yes
extensions.conf
codice:
[mycontext]


exten => 200,1,Dial(dahdi/1/outgoing_number) // dial 200 to dialout from dahdi channel 1
exten => 200,1,Set(LANGUAGE()=it)
exten => 200,2,Hangup

exten => 1001,1,Dial(SIP/1001,10,t,m)
exten => 1001,3,Hangup

exten => 1002,1,Dial(SIP/1002,10,t,m)
exten => 1002,3,Hangup

exten => 1003,1,Dial(SIP/1003,10,t,m)
exten => 1003,3,Hangup

exten => 1004,1,Dial(SIP/1004,10,t,m)
exten => 1004,3,Hangup

exten => 7500,1,VoicemailMain(@mycontext)

exten => 600,1,Answer()
exten => 600,2,Playback(demo-echotest) ; Let them know what
exten => 600,3,Echo()                  ; Do the echo test
exten => 600,4,Playback(demo-echodone) ; Let them know it 
exten => 600,5,Hangup()

exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)        ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)                ; Return to the start over message.

[from-pstn]
exten => s,1,Answer()
exten => s,2,GotoIf(${BLACKLIST()}?blacklisted)
exten => s,3,Dial(SIP/1003&SIP/1002&SIP/1001&dahdi/2,150,t,m)
exten => s,n(blacklisted),Hangup()
exten => s,4,Hangup()
Grazie anticipato a chi mi aiuterà