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--- vendors:cisco:wxcc:local-gateway [2025/05/12 15:59] – created gerardorourke
+++ vendors:cisco:wxcc:local-gateway [2025/05/13 14:12] (current) – gerardorourke
@@ Line 1: / Line 1: @@
-====== Webex Calling - vCUBE Local Gateway Template ======
+====== Webex Calling - vCUBE Local Gateway Config ======
   *Reference: https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling
+  * Control Hub -> Calling -> Call Routing
+===== Registration Based CUBE =====
 ^ Trunk Info                ^ Template Variable     ^ Example Value                                      ^
-| Webex Calling             |                       | Trunk type: Registration Based                     |
 | Line/Port                 |                       | [email protected]  |
 | Line/Port without Domain  | %WX-LINE-USER%        | vCUBE.GerryTest555512345_LGU                       |
@@ Line 14: / Line 16: @@
+==== Sample Config ====
+Replace the variable above with the actual value for your Trunk
+Also replace %WX-CUBE-INTERFACE% with the necessary CUBE interface - e.g. Loopback0 or GigabitEthernet1 etc.
+<code>
+!
+license boot level network-essentials addon dna-essentials
+! SAVE & RELOAD AFTER THIS COMMAND
+!
+ip name-server vrf MGMT 1.1.1.1 1.0.0.1
+ip name-server 1.1.1.1 1.0.0.1
+!
+voice class uri 100 sip
+ pattern dtg=%WX-OTG-DTG%
+!
+key config-key password-encrypt <PASSWORD>
+password encryption aes
+!
+crypto pki trustpoint EmptyTP
+ revocation-check none
+!
+sip-ua
+ timers connection establish tls 5
+ transport tcp tls v1.2
+ crypto signaling default trustpoint EmptyTP cn-san-validate server
+ tcp-retry 1000
+ip http client source-interface %WX-CUBE-INTERFACE%
+crypto pki trustpool import clean url https://www.cisco.com/security/pki/trs/ios_core.p7b
+!
+voice service voip
+ ip address trusted list
+  ipv4 0.0.0.0 0.0.0.0
+ mode border-element
+! SAVE & RELOAD AFTER THIS COMMAND
+ media statistics
+ media bulk-stats
+ allow-connections sip to sip
+ no supplementary-service sip refer
+ stun
+  stun flowdata agent-id 1 boot-count 4
+  stun flowdata shared-secret 0 Password123$
+ sip
+  asymmetric payload full
+  early-offer forced
+!
+voice class srtp-crypto 100
+ crypto 1 AES_CM_128_HMAC_SHA1_80
+!
+voice class codec 100
+ codec preference 1 g711alaw
+ codec preference 2 g711ulaw
+!
+voice class stun-usage 100
+ stun usage firewall-traversal flowdata
+ stun usage ice lite
+!
+voice class uri 100 sip
+ pattern dtg=%WX-OTG-DTG%
+!
+voice class sip-profiles 100
+ rule 10 request ANY sip-header SIP-Req-URI modify "sips:" "sip:"
+ rule 20 request ANY sip-header To modify "<sips:" "<sip:"
+ rule 30 request ANY sip-header From modify "<sips:" "<sip:"
+ rule 40 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
+ rule 50 response ANY sip-header To modify "<sips:" "<sip:"
+ rule 60 response ANY sip-header From modify "<sips:" "<sip:"
+ rule 70 response ANY sip-header Contact modify "<sips:" "<sip:"
+ rule 80 request ANY sip-header From modify ">" ";otg=%WX-OTG-DTG%>"
+ rule 90 request ANY sip-header P-Asserted-Identity modify "sips:" "sip:"
+!
+voice class tenant 100
+  registrar dns:%WX-REGISTER-DOMAIN% scheme sips expires 240 refresh-ratio 50 tcp tls
+  credentials number %WX-LINE-USER% username %WX-CALL-USER% password 0 %WX-CALL-PW% realm BroadWorks
+  authentication username %WX-CALL-USER% password 0 %WX-CALL-PW% realm BroadWorks
+  authentication username %WX-CALL-USER% password 0 %WX-CALL-PW% realm %WX-REGISTER-DOMAIN%
+  no remote-party-id
+  sip-server dns:%WX-REGISTER-DOMAIN%
+  connection-reuse
+  srtp-crypto 100
+  session transport tcp tls
+  no session refresh
+  url sips
+  error-passthru
+  rel1xx disable
+  asserted-id pai
+  bind control source-interface %WX-CUBE-INTERFACE%
+  bind media source-interface %WX-CUBE-INTERFACE%
+  no pass-thru content custom-sdp
+  sip-profiles 100
+  outbound-proxy dns:%WX-OB-PROXY-ADDR%
+  privacy-policy passthru
+dial-peer voice 100 voip
+ description Inbound/Outbound Webex Calling
+ max-conn 250
+ destination-pattern BAD.BAD
+ session protocol sipv2
+ session target sip-server
+ incoming uri request 100
+ voice-class codec 100
+ dtmf-relay rtp-nte
+ voice-class stun-usage 100
+ no voice-class sip localhost
+ voice-class sip tenant 100
+ srtp
+ no vad
+</code>
+Once above config has been added - Check with :
+<code>
+show sip-ua register status
+</code>
+to confirm if the SIP trunk has registered.
+You can also refresh the trunk in control hub and confirm if the trunk is now "online".
+The below config configures the "SIP PSTN Side"
+  * %PSTN-IP-ADDR%
+  * %PSTN-INTERFACE%
+<code>
+!
+voice class uri 200 sip
+  host ipv4:%PSTN-IP-ADDR%
+!
+!
+dial-peer voice 200 voip
+ description Inbound/Outbound IP PSTN trunk
+ destination-pattern BAD.BAD
+ session protocol sipv2
+ session target ipv4:%PSTN-IP-ADDR%
+ incoming uri via 200
+ voice-class sip asserted-id pai
+ voice-class sip bind control source-interface %PSTN-INTERFACE%
+ voice-class sip bind media source-interface  %PSTN-INTERFACE%
+ voice-class codec 100
+ dtmf-relay rtp-nte
+ no vad
+!
+voice class dpg 100
+ description Route calls to Webex Calling
+ dial-peer 100
+voice class dpg 200
+ description Route calls to PSTN
+ dial-peer 200
+!
+dial-peer voice 100
+ destination dpg 200
+dial-peer voice 200
+ destination dpg 100
+!
+</code>