un'ultima domanda (torniamo alle basi)...
quali sono le parti fondamentali
e minime per funzionare il voip, SIP per la precisione, su un router?
voice service pots
voice service voip
voice class codec 1
Codice: Seleziona tutto
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice class codec 2
Codice: Seleziona tutto
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice translation-rule 1
voice translation-rule 2
voice translation-profile SIP
voice translation-profile SIP-IN
voice-port 1/1/1
Codice: Seleziona tutto
ring cadence pattern01
echo-cancel coverage 32
no vad
cptone IT
timeouts initial 5
timeouts interdigit 5
timeouts call-disconnect 5
timeouts wait-release 5
description "0*******"
music-threshold -70
bearer-cap Speech
caller-id enable
dial-peer voice 1 voip
Codice: Seleziona tutto
translation-profile incoming SIP-IN
translation-profile outgoing SIP
destination-pattern .T
session protocol sipv2
session target ipv4:212.97.59.76:5061
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2 pots
sip-ua
Codice: Seleziona tutto
authentication username ******* password *********
disable-early-media 180
retry invite 2
retry response 2
retry bye 2
retry cancel 2
retry register 10
timers connect 1000
registrar ipv4:212.97.59.76:5061 expires 3600
no suspend-resume
che manca/c'è di troppo?