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?