Do you have any ideas suggestions to get this to work with the ISR as a SIP UA?

I don’t think the CM has a way of knowing when PRIs are down. It only sees the connection the the gateway and unless the gateway itself is down, it will continue to send calls to it. This is why I still use MGCP for our gateways. MGCP ensures that when any of the PRI’s are unavailable, the call routes to the next available gateway.

Hi Angelo,
Thanks for the feedback, it’s much appreciated 🙂

It actually seems like you have 99% of this already figured out.
It’s just a matter of avoiding loops between the dial-peers and this is done with a solid numbering plan.

The problem here is that 100 and 102 are conflicting.
I would place the destination pattern 9T on dp 11 instead of 101,
add another voip dp (similar to 11) for ….
on DP 102 replace destination patter with Incoming Called Number …..
and change the destination pattern on 100 to …..

So in this scenario, a call from SME to PBX:
DP 11 –> DP 100

A call from PBX to SME
DP 102 –> DP 11 (+ another DP for 5 digit dial)

If there is still a conflict (loop) between 100 and 102, you can just add a prefix on the incoming DP from the PBX and remove it in CUCM or a VOIP DP.

Hope this makes sense.

What you are publishing is a great content. I love how specific it is to the target audience. I’m hoping that you could help me out with configuring dial-peers with best practice in mind as well as ease to implement across multiple sites. (scalability)

Here is the scenario:

We have multiple sites that still have Nortel PBX equipment we are working on a project that will eventually port all of the carrier PRI DID’s to a IPSTN SIP connection from a single carrier, the SBC’s are in place a few of the major sites have been converted. In doing these sites I’m finding weird behavior with the Nortel (propriety) SIP interoperating with standards-based SIP (Cisco) and would like to move to a more uniform architecture.

Carrier SIPSBCCisco SME(CUCM)SIP to IOS GatewayISDN(PRI)Nortel PBX

The ISDN interaction with the Nortel equipment works beautifully as this is what they built their platform around.

What I would like to do is make a configuration as vanilla as possible, basically my goal is to leave all the routing and heavy lifting to be done by my SME cluster, with the IOS gateway solely serving the purpose of a media converter from SIP to ISDN.

My thoughts were something along the lines of:
Incoming Dial-peer from SME
voice class uri 100 SIP
host ipv4:SME-SUB1
host ipv4:SME-PUB

dial-peer voice 10 voip
description incoming calls from SME
session protocol sipv2
incoming uri via SME
voice-class codec 1
dtmf-relay sip-kpml rtp-nte
no vad

Outgoing calls to SME
dial-peer voice 11 voip
description outgoing calls to SME
session protocol sipv2
session server-group 1 (sme servers)
voice-class codec 1 voice-class sip options-keepalive profile 1
dtmf-relay sip-kpml rtp-nte
no vad

the POTS dial peers I’m not so sure on: really what I’m after is very basic and easy dial peers that can be applied in any situation, that do this….

Anything coming in matching my incoming voip dial peer should be sent to the PRI trunk group. And anything coming in form the PRI trunk group should be sent to the outgoing voip dial peer. Since we have 5-digit extensions across the enterprise I was hoping something like this would work.

Dial-peer voice 100 pots (incoming)
Trunkgroup
Description calls from SME to PBX
Destination-pattern 21[7-9]..
No digit-strip

Dial-peer voice 101 pots (outgoing)
Trunkgroup
Description calls form PBX to IPSTN via SME
Destination-pattern 9T
No digit strip

Dial-peer voice 102 pots (outgoing)
Trunkgroup
Description calls form PBX to SME for internal calls
Destination-pattern …..
No digit strip

My goal ultimately is to not have to define the extension range for a site in the pots dial (100) and make it completely universal and easy to deploy. Then make all routing decisions in the SME rather than the IOS gateway.

Any help is more than appreciated.

Cheers,

Angelo

Hi Pasha,
No worries,
I have created new post under the member’s area 🙂
thank you again.
regards
Paul

Hi Paul,
unfortunately I’m unable to move comments to forum topics, although not a bad idea for a feature 🙂
You’ll have to register and add a topic in the relevant section.

Hi Pasha,
thanks for this,
I’ve been trying to find the post where it has been moved too with out success.
thanks
Regards
Paul

show run
Building configuration…

Current configuration : 3413 bytes
!
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
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!
hostname lavvgw002
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boot-start-marker
boot-end-marker
!
!
enable secret 5 $1$RNqx$fvOhNq/wy1BONvI3pN5GH/
!
no aaa new-model
!
!
!
!
!
!
!
!
!
!
!
!
!
!
ip domain name xx.com.au
ip name-server 10.3.1.81
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
cts logging verbose
!
!
voice-card 0
!
!
!
voice service voip
ip address trusted list
ipv4 10.0.0.0 255.0.0.0
ipv4 210.193.x.x
ipv4 210.193.x.x
address-hiding
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
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modem passthrough nse codec g711alaw
h323
h225 display-ie ccm-compatible
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call start slow
sip
bind control source-interface GigabitEthernet0/0.20
bind media source-interface GigabitEthernet0/0.20
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early-offer forced
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pass-thru content sdp
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voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
!
!
voice class sip-profiles 1
request CANCEL sip-header Max-Forwards modify “.*” “User-Agent: CUBE”
!
!
!
!
!
!
!
license udi pid CISCO2901/K9 sn FGL191422HV
hw-module pvdm 0/0
!
!
!
username pepperxxx privilege 15 secret 5 $1$ptE9$H7kDBpjiokAhzq.OmwaI3/
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 10.3.20.6 255.255.255.0
ip pim dense-mode
h323-gateway voip interface
!
interface GigabitEthernet0/1
ip address dhcp
duplex auto
speed auto
!
ip default-gateway 10.3.20.1
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.3.20.1
!
!
!
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control-plane
!
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 1 voip
description *** Outgoing call . outgoing call leg from CUBE to Broadsoft***
destination-pattern .T
session protocol sipv2
session target ipv4:210.193.xx.xx
voice-class codec 1
no vad
!
dial-peer voice 2 voip
description *** CUBE to UC***
huntstop
answer-address 0391015981.
destination-pattern 0391015981.
media forking
session protocol sipv2
session target ipv4:10.3.20.4
incoming called-number 0391015981.
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
credentials username 039101xxx password 7 0526352611781C50495541425B5E realm domain.com
retry invite 2
retry response 3
retry register 4
timers expires 300000
registrar dns:domain.com expires 120
host-registrar
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!
!
gatekeeper
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!
!
!
line con 0
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line 2
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stopbits 1
line vty 0 4
login local
transport input telnet ssh
!