sip.conf
[trunks](!) type=user insecure=port,invite dtmf=auto context=trunk disallow=all allow=alaw nat=no qualify=yes canreinvite=yes [RtcCiscoGW](trunks) context=pstn-incoming host=10.0.10.250 dtmfmode=auto [038*](RtcCiscoGW) host=dynamic [095*](RtcCiscoGW) host=dynamic
extensions.conf
[pstn-incoming]
exten => _X.,1,NoOp(Date ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => _X.,n,NoOp(Entrant ${EXTEN})
exten => _X.,n,NoOp(CallerId ${CALLERID(all)})
exten => _X.,n,Wait(1)
exten => _X.,n,Dial(SIP/tel00&SIP/c450ip00&SIP/andros00)
exten => _X.,n,Hangup()
[outgoing]
; Ici on sort directement par la FreeBox, ca coute moins cher :)
; Et surtout, on n'a pas l'impression qu'il y a un PaBx :)
exten => _0X.,1,NoOp(Date ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => _0X.,n,NoOp(${EXTEN})
exten => _0X.,n,Dial(SIP/095*/095${EXTEN})
Ce sont des bribrs de confs !!!! lu quelque part qu'il faut remplir le cisco de ces cartes vic de la droite vers la gauche
voice rtp send-recv ! voice service voip allow-connections sip to sip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip registrar server expires max 600 min 60
voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 codec preference 4 g726r32
logging trap debugging logging 10.0.10.5 snmp-server community public RO snmp-server enable traps tty
voice-port 3/0 translate calling 1 translate called 1 supervisory disconnect dualtone mid-call no battery-reversal groundstart auto-tip delay 1000 input gain 9 output attenuation -3 cptone FR timeouts call-disconnect 5 timeouts wait-release 5 connection plar opx 11 impedance complex2 description Ligne FT caller-id enable ! voice-port 3/1 translate calling 1 translate called 1 supervisory disconnect dualtone mid-call no battery-reversal input gain 9 output attenuation -3 cptone FR timeouts call-disconnect 5 timeouts wait-release 5 connection plar opx 12 impedance complex2 description Ligne Free caller-id enable ! voice-port 3/2 shutdown ! voice-port 3/3 shutdown
dspfarm confbridge maximum sessions 1 dspfarm rtp timeout 60 dspfarm codec g729 vad disable dspfarm connection interval 60 dspfarm
dial-peer voice 11 voip huntstop destination-pattern 11 voice-class codec 1 session protocol sipv2 session target ipv4:10.0.10.5:5060 session transport udp dtmf-relay rtp-nte no vad ! dial-peer voice 12 voip huntstop destination-pattern 12 voice-class codec 1 session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte
dial-peer voice 456 pots destination-pattern 095T port 3/1 forward-digits 3 ! dial-peer voice 123 pots destination-pattern 038T port 3/0 forward-digits 3
sip-ua nat symmetric check-media-src max-forwards 4 retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 registrar ipv4:10.0.10.5 expires 3600 sip-server ipv4:10.0.10.5:5060
telephony-service ip source-address 10.0.10.250 port 5060 max-conferences 4 gain -6