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
sip.confcodice:[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
extensions.confcodice:[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
Grazie anticipato a chi mi aiuteràcodice:[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()