 role active R1-PBX(config-sip-ua)# nat symmetric check-media-src R1-PBX(config-sip-ua)# connection-reuse The connection- - reuse sip-ua command is hidden in every version of IOS I've used (up to and including version 15.0(1)XA5. Here's the entire sip-ua configuration section for reference: Small-Office VolP Example sip-ua credentials number 608222222 username 608222222 password 7 023510541B4B3B294D5A48 realm voip.sip-provider.com credentials number 557333333 username 557333333 password 7 023510541B4B3B294D5A48 realm voip.sip-provider.com authentication username 608222222 password 7 023510541B4B3B294D5A48 nat symmetric role active nat symmetric check-media-src no remote-party-id no redirection retry invite 2 retry response 3 retry bye 2 retry cancel 2 retry register 10 timers connect 100 timers register 250 registrar dns:sbc.sip-provider.com expires 3600 sip-server dns:sbc.sip-provider.com connection-reuse host-registrar Now that our SIP trunks are configured, we need to configure the router to accept and send calls. We will need to create two VoIP dial peers. First, we need to do some prep work. I'd like all my calls to use the G.711 CODEC, because I'm a sucker for good sound quality. If that's not allowed, I'll allow G.729r8 to be used. I need to configure a voice class with my preferences. I'll then reference this class in my dial peer: R1-PBX(config-dial-peer)#voice class codec 1 R1-PBX(config-class)# codec preference 1 g711ulaw R1-PBX(config-class)# codec preference 2 g729r8 Now I can build my incoming SIP dial peer: R1-PBX(config)#dial-peer voice 89 voip R1-PBX(config-dial-peer)# description [ Incoming SIP Call ] We need to tell the dial peer that SIP is the protocol in use and that calls matching should be sent to the SIP server, which was configured in the sip-ua section earlier. This may seem odd, since the call is incoming, but the SIP server on the router will know how to transcode the call to H323 SO it can be sent to our Cisco phones: R1-PBX(config-dial-peer)# session protocol sipv2 R1-PBX(config-dia