The scenario is as follows:
Two routers each with a PRI connected, the Telco had
no restrictions as to which PRI could be used for voice calls into the site. Outbound calls would prefer the active CME router, with overflow being sent to the non-active CME router. I have tried to show the requirements in the following diagram.
If we break the configuration down, between the Telephony service, and H323 routing components, the configuration to meet the requirements is pretty straight forward.
Telephony service.
Pre version 4.0 CME redundancy was achieved by using HSRP between the two CME routers, however we can now configure the standby router address directly, this will then assign two "callmanager" addresses to the IP phones (this is similar to a CallManager group if we were using CUCM). In the device menu on the phone the addresses will be listed, with an indication showing which address is the current active address.
To do this the following command is entered on the router that will be the primary CME (as per the address listed as DHCP option 150 :
ip source 192.168.1.1 port 2000 secondary 192.168.1.2
on the secondary router, you can get by with listing only one address:
ip source 192.168.1.2 port 2000
At this point both routers will now be running the Telephony Service, and will accept phone registrations. The rest of the Telephony service will now be identical between the two routers. All call routing will be done via the Dial-peer configurations.
Outbound calls:
For simplification only one dial-peer will be configured.
The following dial-peer will send all traffic destined for 9T (using 9 as the PSTN access number) to the local PSTN voice-port. This dial-peer will be configured exactly the same on on both routers
dial-peer voice 9 pots
destination-pattern 9.T
port 0/0/0:23
If this port is unavailable for some reason due to either a fault, or no available channels, all outbound traffic should be sent to the standby router. Being a VOIP dial-peer all matched digits will be forwarded. The call will then match the PSTN dial-peer that has been configured on the standby router, as mentioned above.
dial-peer voice 91 voip
destination-pattern 9.T
preference 2
session target ipv4:192.168.1.2
Inbound Calls:
Inbound calls require a slightly trickier configuration, as we have to make sure that calls that are received on the standby router are forwarded to the active router, even though the standby router has the same ephone-dn's configured on it as the active router.
There a several ways this can be achieved, the first is to create a dial-peer for each DN and give this dial-peer as better preference than that of the ephone-dn's. Though this will work, it involves quite a bit of extra configuration should new ephones be added.
The method I decided on was to create two dial-peers that matched the digits received from the telco, the most preferred dial-peer will forward the call to the active router, without any translation. The second less preferred dial-peer will match the digits received from the telco, translate the received digits to those of the ephones, and then have itself configured as the dial-peer. This way inbound calls will only be received by the standby router destined for the ephone-dn's them selves if the active router is unavailable.
Hopefully the following configuration will "clarify" things.
First as this solution means that we cannot use the dial-plan configuration command, an inbound translation rule will need to be defined. For our example, the full e164 number is 450-2300 - 450-2399 mapping to extns 1300-1399
voice translation-rule 1
rule 1 /\^45023\(..$\)/ /1\1/
voice translation-profile pstnIN
translate called 1
This translation will need to be configured on both routers.
On the active router, the translation profile is applied to the default inbound pots dial-peer, and the default inbound voip dial-peer (this is required as the standby router is going to forward all digits)
dial-peer voice 1 pots
direct-inward-dial
incoming called-number .
translation-profile incoming pstnIN
port 0/0/0:23
dial-peer voice 1 voip
incoming called-number .
translation-profile incoming pstnIN
codec g711ulaw
no vad
On the standby router, similar dial-peers are configures, however this time the translation profile is not applied to the default pots dial-peer. Tow additional dial-peers are added to forward the inbound PSTN traffic to the active router as first preference, then back to itself as second preference, the number is then translated at this point to match the numbers that have been assigned to the ephones.
dial-peer voice 1 pots
direct-inward-dial
incoming called-number .
port 0/0/0:23
dial-peer voice 1 voip
incoming called-number .
translation-profile incoming pstnIN
codec g711ulaw
no vad
dial-peer voice 91 voip
destination-pattern 45023..
session target ipv4:192.168.1.1
dial-peer voice 91 voip
destination-pattern 45023...
preference 2
translation-profile outgoing pstnIN
session target ipv4:192.168.1.2
That is pretty much it.
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1 comment:
How would this work if you had two CME routers in two physical locations both with their own PRI that were connected together over fiber, and you wanted the phones at each site to prefer their local CME for outbound calls unless it went down?
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