Are you only allowed to make national calls in India via the circuit?
If so you could use the route pattern in CM to drop the 0 (zero) and prefix the called number with +91 en-route to the gateway. Then on the gateway you can use +91T as the destination pattern on the outbound dial peer. As an option you could use 0T as outlined before and use a voice translation rule to rewrite the called number to start with +91 and drop the 0 (zero).
You'll likely also need to modify your calling number information sent to the service provider to fit their requirement. You can do that in a few different ways. Either with a voice translation rule in the gateway or by modifying it in CM via the options that are present on the route pattern.
Dial tone has no correlation with discard digits, that’s controlled by Provide Outside Dial Tone setting on the route pattern.
With the setting that you have in the screenshots you would prefix +91 in front of the zero and that would result in an invalid number as there are very few countries that has a zero as part of the phone number after the country code. The only example of this that comes to mind is Italy, where they keep the leading zero in the area code for inbound international calls.
My question about what calls you should be able to make was possibly somewhat unclear. What I meant are you only allowed by the service provider, once you get the calls to work, to make calls to external numbers in India (+91)?
Okay, thanks for clarification. Then you’re destination pattern should be +T to allow calls to any country and you’ll need to format the sent called number for calls from your CM to the gateway to match this destination pattern.
Can you please share the output from these two debugs for an outbound call?
- debug voip ccapi inout
- debug ccsip message
You have multiple issues at hand.
First off your call is matching dial peer 0 (zero) for the inbound direction from CM to the gateway. This is not good and should be avoided by having a specific configured dial peer that is used for this direction. Previous I gave you the advice to use information in the via header for inbound dial peer match. Please configure this, you’ll need two inbound dial peers, one from CM and one from your service provider.
Secondly you are matching dial peer 300 for the outbound direction, and that’s good as that’s your intended outbound dial peer towards your service provider. However your destination pattern is +91, not +T or +91T as has been suggested by me. It’s very hard for anyone to help you if you do not use the advice as given. Please correct this.
Thirdly your sending calling number in the wrong format as per your service providers requirements. Please fix that. To do so use voice translation rules in the gateway. Either inbound from CM or outbound from the SBC (Cube) towards the service provider.
Lastly the call is disconnected with this error.
ccCallDisconnect:
Cause Value=41
This is quite likely caused by either that your using the wrong codec as the call matches dial peer 0 inbound, this dial peer is hard coded for g729, or it could be caused by the wrong number format.
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.
Glad to hear that you got it to work. About your questions.
voice class e164-pattern-map 1
This is used to map out your extensions you have in your CM. Usually and what is more manageable is to have a range of extensions listed in the list. For example like this.
voice class e164-pattern-map 1 description E164 Pattern Map for CUCM numbers e164 8..
For better manageability it would be advisable if you could reuse some of the digits that you get from your service provider as part of your DID range. For example if you would keep the last two digits then your extensions in CM could be 800 and 801. With this you could simplify your inbound and outbound number translations to this.
voice translation-rule 100 rule 1 /.*1\(..\)$/ /8\1/ ! voice translation-rule 200 rule 1 /^8\(..\)$/ /+9126323501\1/
For your second question. This translation rule is used to change the called number outbound from CM from starting with 0 and then more digits to start with +.
voice translation-rule 110 rule 1 /^000\(.*\)/ /+\1/
rule 2 /^0\(.*\)/ /+91\1/
Rule 1 match on three zeros and copies the rest of the digits into memory and prefix a + in front.
Rule 2 match on one zero and copies the rest of the digits into memory and prefix +91 in front.
