In the preceding seven steps, you set up the entire Cisco IPC Express system at Site A with examples of the most commonly deployed features you will require for your office. Although the Site B configuration was not discussed step by step, its configuration is very similar to that for Site A, with extensions starting at 3001 instead of 2001.
You cannot yet make calls between Sites A and B, because there is no dial plan to route calls between the sites. To achieve this, first ensure IP routing between the sites. In the sample configuration being built in this chapter, Site A has an IP address of 10.1.235.1 (with a netmask of 255.255.0.0), and Site B has an IP address of 10.1.229.1 (with a netmask of 255.255.0.0), so these systems can easily reach each other. If your sites' IP addressing is more sophisticated, do the necessary configuration to achieve IP routing between your sites. Ensure that you see routes between the IP addresses of your sites with a show ip route command on each site's router. You should also be able to ping one site from the other.
Interconnecting Sites Via H.323
As soon as you have IP connectivity between the sites, the next step is to add dial peers to route calls between the sites. From Site A, if someone dials an extension that starts with 3, the call must be routed to Site B. Similarly, if someone at Site B dials an extension starting with 2, the call must be routed to Site A. If your dialing plan is less uniform than the sample network in this chapter, you may need multiple dial peers to route all calls. Also, if only one site has PSTN access, and DID numbers for both the 2xxx and 3xxx ranges arrive on one PSTN trunk, more dial peers are needed to route all calls correctly.
For the sample network, the dial peers to route calls between the sites are shown in Example 15-49.
Example 15-49. Sites A and B Dial Peers
! Site A (2xxx extension) dial-peers to direct calls to Site B (3xxx extensions) cme-3725#show running-config dial-peer voice 3000 voip destination-pattern 3... session target ipv4:10.1.229.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ! Site B (3xxx extension) dial-peers to direct calls to Site A (2xxx extensions) cme-2691#show running-config dial-peer voice 2000 voip destination-pattern 2... session target ipv4:10.1.235.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad
At this point, the sites can call each other by simply dialing the extensions of the IP phones. One more configuration must be added to ensure that you can transfer calls between the sites. Add the transfer-patterns shown in Example 15-50 to both sites' configurations.
Example 15-50. Sites A and B Transfer Patterns
cme-3725#show running-config telephony-service transfer-system full-consult transfer-pattern 3... transfer-pattern 2...
In the preceding configuration setup, G.711 is used for all calls, including those between sites. To conserve bandwidth on the link between your sites, it is likely that you want to use G.729 on those calls instead. If so, remove the codec g711ulaw statement from the dial peers in Example 15-49.
You can specify G.729 explicitly (by using the codec g729r8 command), or you can simply delete the G.711 statement, because G.729 is the default codec for a VoIP dial peer. Whether the actual codec used is G.729 or G.729A depends on the codec complexity configuration of the PSTN trunk voice card. It doesn't matter for call connectivity, because G.729 and G.729A are fully compatible with each other.
Transcoding is required when part of a call must use the G.711 and another part of the same call must use G.729. When you use G.729 for calls between sites, and calls forward into voice mail, these calls currently fail on the configuration, because Cisco UE voice mail supports only G.711. To fix this, configure transcoding resources on both sites to terminate G.729 calls, and transcode them locally to G.711 before they enter voice mail.
Example 15-51 gives a sample transcoding configuration. Ensure that you have enough digital signal processor (DSP) resources on the voice cards in your system to support this. If you don't, add more DSPs.
Example 15-51. Transcoding
router#show running-config voice-card 2 dsp services dspfarm ! interface Loopback1 ip address 10.32.153.45 255.255.255.252 h323-gateway voip interface h323-gateway voip bind srcaddr 10.32.153.45 ! sccp local Loopback1 sccp ccm 10.32.153.45 identifier 1 sccp sccp ccm group 1 bind interface Loopback1 associate ccm 1 priority 1 associate profile 1 register MTP000e833595e0 keepalive retries 5 dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec gsmfr codec g729br8 codec g729r8 maximum sessions 10 associate application SCCP telephony-service ip source-address 10.32.153.45 port 2000 sdspfarm units 1 sdspfarm transcode sessions 30 sdspfarm tag 1 MTP000e833595e0
Transcoding is required to support the following call flows:
SIP RFC 2833 DTMF Relay
You need SIP DTMF relay if you are using SIP trunking between sites. If you have Cisco UE integrated on your sites, as in the sample configurations built in this chapter, you must use H.323 trunking between the sites. SIP trunking is not yet supported with Cisco UE release 2.1. Note, however, that a SIP dial peer is required to route IP phone and PSTN calls to Cisco UE.
If you are using AA and voice mail solutions other than Cisco UE with Cisco CME, or a future Cisco UE software release that may support this feature, you can use SIP trunking between sites. In a SIP trunking configuration, the out-of-band DTMF relay to the SCCP IP phones must be converted to in-band RFC 2833 DTMF relay on the SIP trunk. This is done using the configuration sample shown in Example 15-52.
Example 15-52. DTMF Relay for SIP Trunking
router#show running-config dial-peer voice 2000 voip destination-pattern 8005551212 session protocol sipv2 session target ipv4:18.104.22.168 dtmf-relay rtp-nte
Sample System Configurations