09-26-2008 12:01 PM - edited 03-15-2019 01:34 PM
I'm trying to configure dial peers on 2821 that will be a CUBE gateway device for a CUCM 6.0 server. The router is connected to an MPLS network and the link is configured as a SIP trunk on the Telco's site.
I've attached a hypothetical network diagram to simplify things. What would be the best way for the dial peers to work in this configuration, assuming that the incoming DID range was sayâ¦555-555-1250 to 555-555-1350?
I have reviewed this document: http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml#configs
I'm getting confused with the inbound/outbound dial-peers. Telco is sending 10 digits and CUCM is sending 10 digits. In the about doc it uses 8... as the dial pattern but I'm confused on how the CUBE gateway will differentiate the calls and send them to the correct destinations.
If anyone can shed some light on this I would greatly appreciate it.
Thanks
09-26-2008 05:35 PM
hi Gregory
ket me describe the dial-peers logic for u
in a dial-peer lets say in ur case VOIP dial-peer
the incoming called-number mean this dial-peer will look at the incomming called number thus this is for inbound calles
if the teleco sedn u all calls every number then ur command nay be looks like
incoming called-number .
session target ipv4:[cucm ip]
the . with incoming called-number mean this will be the defual inbound dial-peer will match every number coming to ur gateway
this is in call legs the incoming call leg
while with outgoing the number will look for outbound dia-peer this way the dial-peer with destination pattren will takeplace
lets consider ur case
dial-peer voice 9 voip
desecrition outgount to ur teleco
destination pattren 9.........
session target ipv4:192.168.1.1
session protocol sipv2
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 91 voip
description inbound to cucm
incoming called-number 5555551[23]..
session target ipv4:10.10.1.2
and the following link will help u understand call legs with dial-peer
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
good luck
if helpful Rate
12-12-2008 11:30 AM
Hello, I have a similar issue where we have a Nat Router natting calls from various CM servers and phones on various subnets to the mpls router out to the ATT IPGW. We can not get any more ATT IP addresses to nat with so we want to set up a cube router with NO gatekeeper as an H323 Gateway in CM and route calls to it. I could use help with the config for this. I was going to use both eth interfaces with media pass-through. How does the Cube work, does it NAT?? I am a bit lost. I have looked at ALL docs on Cisco website every example they have, I have looked at...
Thansk,
12-12-2008 12:34 PM
First you have to understand dial peer selection:
1. Incoming called-number
2. Answer address
3. Destination Pattern
4. Port
First off, 4 doesn't apply because there are no analog ports.
Every call matches an incoming and outgoing dial peer.
You could match the incoming dial peer for your SIP provider on the incoming called-number of your CUCM phones.
You would match the outgoing dial peer for incoming SIP calls with a destination-pattern towards CUCM.
For incoming CUCM calls you could match an incoming dial peer based on answer-address (your DID range)
For incoming CUCM calls you can match the outgoing dial peer on a wildcard dial peer such as any 9/10/11 digit number with a destination pattern.
Ex:
dial-peer voice 1 voip
desc incoming SIP dial peer
incoming called-number 55555512[5-9].
codec g711ulaw
no vad
dtmf rtp-nte
dial-peer voice 2 voip
desc outgoing cucm dial peer (for SIP-SIP CUBE)
destination-pattern 55555512[5-9].
session target ipv4:10.10.1.2
session protocol sip
dial-peer voice 21 voip
answer-address 55555512[5-9].
desc incoming dial peer for CUCM
dial-peer voice 11 voip
desc outgoing dial peer to SIP trunk
destination-pattern .T
session target 192.168.1.1
session protocol sip
In this case the settings are you codec, DTMF, no vad, translation patterns, etc.
This is for a SIP-SIP call. Please note you shouldn't use the SIP bind command unless both your SIP provider and CUCM will be sending to the same address.
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