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AMA-CUCM Troubleshooting: Best Practices for Reading Trace Files

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Beginner

Dial-Peer Question

I have a new CUCM cluster that I'm taking over for our California region and they are porting some numbers from AT&T to their Level 3 SIP trunk. I want to make sure that when the port happens that the new numbers are able to make calls outgoing and incoming. I spoke with TAC and they told me that from looking at my route patterns the outgoing will work fine. But he informed me that he would have to engage someone from the gateway team to look over my dial peers to configure the correct one for the new numbers to work for incoming. The engineer is currently on vacation this week and the port is happening this Friday. I wanted to know can someone look over these dial-peers and tell me what do I need to add if anything to make these new numbers work for incoming calls. The numbers that I'm speaking of are in the 650365XXXX and 650395XXXX range and thank you for the help and I look forward to your replies.

 

Eric

6 REPLIES 6
Hall of Fame Cisco Employee

Re: Dial-Peer Question

I'd say this is the perfect opportunity for you to review the dial peer configuration guides and learn how to do this, this is not that hard, it's just regex and there's plenty of documentation on how dial peers work (inbound/outbound) and how matching works.

HTH

java

if this helps, please rate
Beginner

Re: Dial-Peer Question

Jamie

I looked over the configuration manual and I wanted to know can you tell me if this is correct for calls coming into CUCM. The numbers that I'm working with are in the 650365XXXX  and 650395XXXX range.

 


dial-peer voice 100 voip
description Incoming calls to CUCM1
preference 1
destination-pattern 650365....
session protocol sipv2
session target ipv4:192.168.248.3
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

dial-peer voice 101 voip
description Incoming calls to CUCM2
preference 1
destination-pattern 650365....
session protocol sipv2
session target ipv4:192.168.248.7
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

dial-peer voice 102 voip
description Incoming calls to CUCM1
preference 1
destination-pattern 650395....
session protocol sipv2
session target ipv4:192.168.248.3
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

dial-peer voice 103 voip
description Incoming calls to CUCM2
preference 1
destination-pattern 650395....
session protocol sipv2
session target ipv4:192.168.248.7
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte
no v

Hall of Fame Cisco Employee

Re: Dial-Peer Question

Participant

Re: Dial-Peer Question

Couple of things to look at, one is E164 pattern maps, allowing each dial peer to match multiple destination patterns.  That saves having to create a new set of CUCM facing dial peers for each new number range. 

Secondly on a SIP gateway I like to use some mechanism to restrict which dial peers can be seen.  For example class of service (corlist) set so that any incoming PSTN call can only see the CUCM dial peers, and a CUCM dial peer can only see the PSTN.  With that in place the number matching becomes less specific as the inbound and outbound calls are essentially separated.  For this to work you need to have bombproof inbound dial peer matching, and for SIP I prefer to match by IP address, again removing some of the number matching requirements.

Enthusiast

Re: Dial-Peer Question

Eric,

In your existing dial peer configuration, i see two patterns:

4084042[0-4]..

+1510.......

You need to decide which one will fit in with your new requirement.

 

Also i noticed that the session target IPs in your existing configuration and your new dial peers are different:

Existing:

CUCM01 -> 192.168.248.1
CUCM02 -> 192.168.248.3

 

New dial peers:

CUCM01 -> 192.168.248.3
CUCM02 -> 192.168.248.7

 

Please review these details also before applying the configuration.

Beginner

Re: Dial-Peer Question

Piyush

I made a mistake when I copied the dial peer information into the post, but i got the dial-peer created yesterday afternoon. This is what was created and now since I have that done I'm going to start reading on how to configure dial peers. I'll going to take my time so I can get a good understanding.

 

voice class server-group 1
ipv4 192.168.248.3 pref 1
ipv4 192.168.248.1 pref 2


dial-peer voice 120 voip
description Incoming PRI calls to CUCM
preference 1
destination-pattern 6503[6,9]5....$
progress_ind setup enable 3
session protocol sipv2
session server-group 1
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

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