SIP dial plan considerations are similar to those for an H.323 gateway, because dial peers control gateway call routing. You must configure dial peers to forward calls out of the gateway. You can forward calls to CallManager using a Voice over Internet Protocol (VoIP) dial peer, to the PSTN as a plain old telephone service (POTS) dial peer, to another gateway using a VoIP dial peer, or to directly connected voice ports as POTS dial peers.
In a voice network that has CallManagers and SIP gateways, it is important to understand the interaction between the two, because different versions of CallManager have different SIP capabilities. CallManager versions before 5.x can only have a SIP trunk to a gateway or other servers. CallManager 5.x and above also acts as a SIP B2BUA and allows SIP phones to register to it. It also can do domain routing for SIP calls. A new menu, SIP Route Pattern, allows you to configure SIP URI dialing. Therefore, your dial plan must take into account the CallManager version. No matter which version of CallManager you use, you configure a dial plan to send calls to the SIP trunk when needed. CallManager appears to the SIP gateway as a SIP-enabled VoIP dial peer.
Another consideration in SIP networks is where the dial plan will reside. The default behavior of SIP is to push down the dial plan to each endpoint. When a user dials digits on the phone, the phone compares those numbers against its internal dial plan. If the phone finds a match, it sends an INVITE. Otherwise, it must wait for the interdigit timer to expire before playing a reorder tone. The alternative is to use the Key Press Markup Language (KPML). When you use KPML, the SIP phone sends each digit to CallManager, similar to the way SCCP phones behave. CallManager can instruct the phone to play a reorder tone immediately if an incorrect number is dialed, or it can route the call as soon as enough digits are dialed. If you do not use KPML, you must configure SIP dial rules.
In your dial planning, consider the need to configure the gateway for such options as number translations or other digit manipulations, or call restrictions. If you are using SRST, be sure that the dial plan will work both with and without CallManager and, if possible, any SIP servers in the network. You need at least one dial peer with a destination pattern for routing outgoing calls. Default incoming POTS and VoIP dial peers are available, but you should specifically configure dial peers for incoming calls if you need a nondefault configuration.
As with H.323, SIP gateway configuration can become complex in a large network. You must configure each gateway with the information you need to route calls. Proxy, registrar, redirect, and DNS servers can help the network scale by providing dial plan resolution. This simplifies the gateway configuration.
Part I: Voice Gateways and Gatekeepers
Gateways and Gatekeepers
Part II: Gateways
Media Gateway Control Protocol
Session Initiation Protocol
Connecting to the PSTN
Connecting to PBXs
Connecting to an IP WAN
Influencing Path Selection
Configuring Class of Restrictions
SRST and MGCP Gateway Fallback
Using Tcl Scripts and VoiceXML
Part III: Gatekeepers
Part IV: IP-to-IP Gateways
Cisco Multiservice IP-to-IP Gateway
Appendix A. Answers to Chapter-Ending Review Questions