l-peer)# session target sip-server This next line matches any incoming call: R1-PBX(config-dial-peer)# incoming called-number .% This may or may not work for you. In my case, it looked great and everything was fine until I started working with voicemail and MWIs. Since the AIM-CUE uses SIP to send MWI commands, these messages were matched by the SIP dial peer, making all the phones ring anytime a message lamp status was changed. I fixed this by making the called-number command match any number longer than the MWI-configured dial Chapter VolP peer's match. My MWI dial peer is 800[12]..., SO this statement matches anything lon- ger than 7 digits. Since all incoming calls from my SIP provider have full 10-digit num- bers, this works perfectly: R1-PBX(config-dial-peer)# incoming called-number Here's where we reference the voice class that we just configured SO we can support G.711 and G.729: R1-PBX(config-dial-peer)# voice-class codec 1 This command helps translate DTMF tones to named telephone events. On compressed calls (such as when G.729 is used), DTMF tones can become distorted. This command translates DTMF tones into text-based numbers used in SIP packets: R1-PBX(config-dial-peer)# dtmf-relay rtp-nte Voice activity detection (VAD) is also sometimes called silence suppression. With VAD enabled, perceived silence causes the router to stop sending voice packets. While this saves a lot of bandwidth, the resulting pure silence often causes people to think that the call has died. This can be prevented by injecting synthetic comfort noise on the far end to simulate normal background noise. I generally prefer to avoid VAD altogether unless I have a dramatic need for it: R1-PBX(config-dial-peer)# no vad With our incoming dial peer configured, theoretically we should now be able to receive calls. In actuality, though, we cannot. Though our SIP trunks are properly registered, any calls to our numbers results in a "number not available" message from our provider. This is maddening if you're new to the whole SIP experience,