CUCM sip CUBE for two service providers

[Bekzod Fakhriddinov]

I found this config in our forum :

interface loopback1----------------------------------------Interface pointing to SIP provider 1
ip address 10.10.10.1 255.255.255.0

 interface loopback2-----------------------------------------Interface pointing to SIP provider 2
ip address 20.20.20.1 255.255.255.0

 dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1

dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2

1 Can anyone explain how to connect to sip Service provider with loopback ip? We usually connect with public ip via mpls..

2  How to separate incoming calls from different SP and how inbound dial-peer will look like at this case ? 

Thank you 

[R0g22]

You can only connect via a loopback if your ITSP will be able to route traffic back to you using your private IP addressing, which is highly unlikely.
Further, you can separate the incoming dial-peers in order to match force the IOS to select the specific dial-peers that have the correct binds. You can either make use of E.164 pattern maps or incoming URI.

[Bekzod Fakhriddinov]

Can old style dial-peer config and uri based dial-peer coexist/sit on the same CUBE and work ?  

[R0g22]

It should yes.

[Bekzod Fakhriddinov]

HI, can u please tell me  what is the algorithm in dial-peers, for example if there is 3 dial-peers to the same destination and dest pattern but different preferences and all of them has translation profile to change  outgoing digits? 

I see on the trace call is hitting 1st dial-peer not changing digits then hitting next dial-peer not changing digits then next dial-peer... I thought changing should happen before to hit second dial-peer because of translation profile is inside of the 1st dial-peer ...  But looks like there is different algorithm ? 

[R0g22]

The higher the preference on the dial-peer, the least preferred it will be. So if you have 3 DP's with same destination pattern but with a preference of 1,2 and 3 respectively, the DP with preference 1 will be used always. Then the DP with pref 2 and the DP with pref 3 will be the least preferred.

[Bekzod Fakhriddinov]

right pref 1 is preferred than prefer 2 , i know that, My question was why translation profile with changing dialed number didn't work when call hit dial-peer 1 ? 

I see cal is hitting 1 dial peer than 2nd than 3rd , Although i had translation profile to delete some digits from dialed number...  Is there any logic I am missing ? 

thank you 

[Bekzod Fakhriddinov]

here is config :

voice translation-profile StripTESTLocalAccessCode
translate called 7
translate redirect-called 7

voice translation-rule 7 ## for outg calls to WCS
rule 1 /^7011\(.*\)/ /011\1/ type international international
rule 2 /^7\(1.*\)/ /\1/ type national national
rule 3 /^7\(.*\)/ /\1/ type subscriber subscriber

dial-peer voice 7 voip
description ** Outgoing Dial-Peer to WCS_NY from CUCM **
translation-profile outgoing StripTESTLocalAccessCode
preference 2
destination-pattern 7T
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 100
voice-class sip profiles 2
voice-class sip options-keepalive up-interval 30
voice-class sip session refresh
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp af41 signaling
no vad

[R0g22]

Can you show dial-peer with preference 1 ? Also, what was the calling and called number for this call for which translation did not work ?

[Bekzod Fakhriddinov]

Nipun you are the best men, u always trying to help. Thank you. I could fix this part right now, I saw  its sending subcriber or national while there was something with unknown and decided to delete them, there was type national,international and subscriber on the translation rules , I deleted types from rules and now calls goes trough. Now its hitting dial-peer 7 then changing it and my cell phone ringing . 

Only thing now i need to figure it out when call connects i have no audio , its probably  because they have different ip for signaling and differnt for media. I have to find out is it routing issue... 

[Bekzod Fakhriddinov]

Nipun, I captured pkacp files from wan and lan interfaces of cube and sii ITSP is sending trying after my invite and my wan is getting it but CUBE is not sending anything to CUCM then to LAN port, neither trying or something else. I have their ip as trusted , i don't understand ... On the wan port i have only policy map for qos allowing ports for signaling and media , so it should not block them. 

Where also I can look on my configuration?  

thank you

[R0g22]

What is this for ? Outbound call, inbound call ? Can you enable ccsip and ccapi debugs and collect a call log ?

[Mohammed al Baqari]

To use your public IP as source, specify the public interface as source in
dial-peer voice class command.

To match calls from different sip providers and use different source
interfaces, use URI matching to match the host field in VIA header. This
is usually the ITSP IP. Once the dialpeer is matched the correct source
interface is invoked.

For example:

voice class uri itsp-01 sip
host ipv4:192.1.20.1
!
voice class uri itsp-02 sip
host ipv4:192.1.10.1
!
dial-peer voice 10 voip
session protocol sipv2
incoming uri via itsp-01
voice-class sip call-route url
voice-class sip source-interface g0/0.10
!
dial-peer voice 11 voip
session protocol sipv2
incoming uri via itsp-02
voice-class sip call-route url
voice-class sip source-interface g0/0.11