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CUBE with multiple SIP to same SP

Hello

 

I need your help and inputs...I need to add "another country" to our CUBE. Our use case is:

Actually, we are service country A (Switzerland) with a SIP Trunk from SP "ABC". Incoming and outgoing calls are working fine.

Between SP "ABC" and our CUBE, we have a SIP Trunk. On SP side, they have one IP for Signaling and one IP For the RTP Stream. We have one DDI range which is forwarded from CUBE to our CUCM.

Now, I have to add a second country (DDI) and for this, I have to speak with another IP for the Signalisation. RTP IP will be the same.

How do I need to configure the CUBE so that DDI of country A (Switzerland) will use the Signalisation IP A and the DDI of country B (Spain) will use the Signalisation IP B? Please check also my quick and dirty attached diagram.

 

This kind of design on the WAN side with this dedicated Signalisation IP's is mandatory from our SP.

In the future, we need to add some more countries and this fact I have to consider as well in the configuration.

 

Some infos about our environment:

- CUCM 11.5.1 SU5

- CUBE ISR4451-X (Cisco IOS XE Software, Version 03.16.05.S)

 

Sorry for my english :-)

 

I hope that anybody can help me. Thank you very much.

 

Best regards,

Pascal

1 REPLY 1
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Participant

Re: CUBE with multiple SIP to same SP

Match dial peers on URI. Here is an example.

voice class uri 404 sip
host voxbone.com
host ipv4:81.201.82.45


voice class uri 401 sip
host colt.com
host ipv4:10.252.7.20

dial-peer voice 404 voip
description Voxbone incoming
translation-profile incoming TRANSLATE
session protocol sipv2
session target sip-server
incoming uri from 404
voice-class codec 1
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/3/1
voice-class sip bind media source-interface GigabitEthernet0/3/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 401 voip
description ATT incoming
translation-profile incoming TRANSLATE
session protocol sipv2
session target sip-server
incoming uri from 401
voice-class codec 1
voice-class sip early-offer forced
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
no vad
voice class uri 401 sip
host colt.com
host ipv4:10.252.7.20

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