Please follow the instructions, in order to configure a modulus SIP trunk with a telephone number on a Cisco CUBE.
Below there is a sample configuration for modulus SIP trunk on Cisco CUBE.
Notes: Comments after hash symbol (#) in the lines below should be omitted
Please replace values in the template notes in square brackets (<abc>) according to your own details (phone number, username, password etc.)
In the sample configuration we have these assumptions:
1. Company uses 0 to perform external calls
2. Company has already configured extension 3103 for Auto-Attendant to route incoming calls
We are creating translation rules and translation profiles to use
voice translation-rule 501 rule 1 /.*/ /<3103>/ #--> 3103 is an arbitrary internal configured extention number for the Auto Attendant
voice translation-rule 502 rule 1 /.*/ /<2152151500>/ #--> telephone number is configured as Caller ID for the outgoing calls (eg. 2152151500)
voice translation-rule 503 rule 1 /^00030/ /<+30>/ #--> removing leading 0 which is used as prefix for outgoing calls rule 2 /^000/ /+/ rule 3 /^0/ //
voice translation-profile MODULUS-IN translate called 501
voice translation-profile MODULUS-OUT translate calling 502 translate called 503
In Cisco CUBE we have to set modulus network as trusted network in order to perform outgoing calls and receive incoming call
voice service voip ip address trusted list ipv4 185.73.40.0 255.255.252.0 #--> modulus network to trust
We define preferred codecs and their priority for SIP trunk modulus
voice class codec 100 codec preference 1 g722-64 codec preference 3 g711alaw codec preference 5 g729r8 codec preference 7 g711ulaw
Continuing by configuring SIP trunk details, using the credentials we have received from modulus regarding endoint activation
voice class tenant 1 registrar dns:voips.modulus.gr expires 180 credentials number <username-as-provided> username <username-as-provided> password 0 <clear-text-password> realm voips.modulus.gr authentication username <username-as-provided> password 0 <clear-text-password> realm voips.modulus.gr retry invite 8 retry response 4 retry bye 8 retry prack 6 retry register 4 sip-server dns:voips.modulus.gr connection-reuse via-port no host-registrar session transport udp bind control source-interface GigabitEthernet2 #--> The interface used for IP connection toward modulus (eg. GigabitEthernet2) bind media source-interface GigabitEthernet2 #--> The interface used for IP connection toward modulus (eg. GigabitEthernet2) no pass-thru content custom-sdp outbound-proxy dns:voips.modulus.gr midcall-signaling block dial-peer voice 1234 voip translation-profile incoming MODULUS-IN translation-profile outgoing MODULUS-OUT huntstop destination-pattern 0T #--> This line is matching dialed zero following any digit sequence during hunting process for outgoing calls session protocol sipv2 session target dns:voips.modulus.gr incoming called-number <2152151500> #--> This line is matching incoming calls that are destined to specific modulus number (π.χ. 2152151500) voice-class codec 100 voice-class sip tenant 1 dtmf-relay rtp-nte no vad sip-ua retry invite 2 connection-reuse tcp-retry 1000 handle-replaces
We can check the status of the trunk, by executing below command, waiting as a result in reg field 'yes'.
#show sip-ua register status Tenant: 1 --------------------- Registrar-Index 1 --------------------- Line peer expires(sec) reg survival P-Associ-URI ================================ ========= ============ === ======== ============ <username-as-provided> -1 72 yes normal --
After the above configuration has been completed, you should be able to perform outgoing calls using prefix 0, as also incoming calls to your telephone number, should be routed to the Auto Attendant 3103.