Accepted Solutions
Knowing that it might be hard to understand all the things mentioned in my prior response I took the time to re-work your configuration to suggest what I would have configured for your described setup. The configuration is based on your previous shared configurations.
voice service voip ip address trusted list no ipv4 100.64.77.89 ipv4 100.64.216.4 ipv4 172.16.20.11 ipv4 172.16.20.9 mode border-element license capacity 20 media bulk-stats allow-connections sip to sip sip header-passing error-passthru early-offer forced midcall-signaling passthru ! no voice class uri 1000 sip
no voice class uri cumctocube sip ! voice class uri CUCM sip host ipv4:172.16.20.11 host ipv4:172.16.20.9 ! voice class uri JIO sip host ipv4:100.64.216.4 ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw ! voice class e164-pattern-map 1 description E164 Pattern Map for CUCM numbers e164 894 e164 826 ! voice class e164-pattern-map 200 description E164 Pattern Map for PSTN numbers e164 +T ! voice class server-group 1 ipv4 172.16.20.11 preference 1 ipv4 172.16.20.9 preference 2 ! voice class server-group 200 ipv4 100.64.216.4 ! voice class sip-options-keepalive 1 ! no voice translation-rule 100 no voice translation-rule 101 no voice translation-rule 200 ! voice translation-rule 100 rule 1 /.*101$/ /894/ rule 2 /.*100$/ /826/ ! voice translation-rule 200 rule 1 /^894$/ /+912632350101/ rule 2 /^826$/ /+912632350100/ ! voice translation-rule 110 rule 1 /^000\(.*\)/ /+\1/
rule 2 /^0\(.*\)/ /+91\1/ ! no voice translation-profile 101 no voice translation-profile TP100 ! voice translation-profile PSTN-IN translate called 100 ! voice translation-profile PSTN-OUT translate calling 200 ! voice translation-profile CUBE-IN translate called 110 ! no dial-peer voice 101 voip no dial-peer voice 201 voip no dial-peer voice 400 voip no dial-peer voice 300 voip no dial-peer voice 200 voip no dial-peer voice 100 voip ! dial-peer voice 1000 voip description ****Inbound CUCM to Cube**** translation-profile incoming CUBE-IN session protocol sipv2 incoming uri via CUCM voice-class sip bind control source-interface GigabitEthernet0/0/0 voice-class sip bind media source-interface GigabitEthernet0/0/0 dtmf-relay rtp-nte sip-kpml voice-class codec 1 no vad ! dial-peer voice 1010 voip description ***Outbound Cube to CUCM*** destination e164-pattern-map 1 session protocol sipv2 session server-group 1 voice-class sip options-keepalive profile 1 voice-class sip bind control source-interface GigabitEthernet0/0/0 voice-class sip bind media source-interface GigabitEthernet0/0/0 dtmf-relay rtp-nte sip-kpml voice-class codec 1 no vad ! dial-peer voice 100 voip description ***Inbound JIO to Cube*** translation-profile incoming PSTN-IN session protocol sipv2 incoming uri via JIO voice-class sip bind control source-interface GigabitEthernet0/1/1 voice-class sip bind media source-interface GigabitEthernet0/1/1 dtmf-relay rtp-nte voice-class codec 1 no vad ! dial-peer voice 110 voip description ***Outbound Cube to JIO**** translation-profile outgoing PSTN-OUT destination e164-pattern-map 200 session protocol sipv2 session server-group 200 voice-class sip options-keepalive profile 1 voice-class sip bind control source-interface GigabitEthernet0/1/1 voice-class sip bind media source-interface GigabitEthernet0/1/1 dtmf-relay rtp-nte voice-class codec 1 no vad !
There might be part(s) that needs a little polishing, but over all this should be a working solution based on the information you have provided. Try it out and let us know how it went.
AFAICT you have no dial peer(s) in the direction of your service provider. Without this your not going to have any SBC functionality in your CUBE. For more details on how to setup an SBC please see this document.
For details of how dial peers works in general please see this document.
The destination number on your outbound dial peer to your service provider does not look correct.
dial-peer voice 300 voip
description ***Outbound Cube to JIO ****
destination-pattern 912632350100
session protocol sipv2
session target ipv4:100.64.216.4
Check this and modify it to match how you send outbound called number from CM to your voice gateway and then onwards to your service provider.
Isn’t that destination number, +912632350100, an inbound number, ie it’s part of your assigned DID number range? You should have your destination pattern set to match the outbound called number(s), not inbound DID number(s). For example if you use +E.164 formatted called number you could use +T as the destination pattern or if you use a breakout code such as 0 or 9 you could use 0T or 9T.
Apart from this I would recommend that you use information in the via header for inbound dial peer matching. This gives you a more deterministic match for your inbound dial peers, one for the call leg from CM and another for the call leg from your service provider. For details on this see the second link that I shared earlier.
I would not recommend you to use a dot as the destination pattern as that’s a to wide match statement. What is your match statement in CM for the outbound calls on your route pattern(s)?
Based on that I would recommend you to use 0T as the destination pattern on the outbound dial peer to your service provider and use a voice translation profile to remove the leading 0 (zero). To do so make sure that you don’t drop the zero on the route pattern in CM when sending the calls to the gateway.
For details on how to use voice translation profile and rules see this document.
https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voice-transla-rules.html
