inging Via: SIP/2.0/UDP 208.73.146.95:5060;branch=z9hG4bK8v1qso00903gmpgqi501.1 From: "DONAHUE GARY sip:+19735551212@192.168.101.112:5060>;tag=gK025eb40 To: <sip:9735554333@192.168.101.100:5060>;tag=3F32A10-2138 Date: Thu, 03 Feb 2011 19:29:03 GMT Call-ID: pcst12967613435153831028210@192.168.201.112 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:133@192.168.0.3>;party=called;screen=no;privacy=off Contact: <sip:608222222@192.168.0.3:5060> Server: isco-SIPGateway/I0S-12.x Content-Length: 0 This output can vary significantly from version to version. When I was using CME v4, the To: <sip: 9735554333 actually changed to <sip: 133... after the translation was applied. Now that we've got calls coming in, let's set up our outgoing dial peers. With your newfound experience, you probably realize that we'll likely have to do translations outbound, too. In my case, if I dial from any extension, the SIP messages will include those extensions in the SIP INVITE and I'll get a "not authorized" message from the provider, because my extensions are not registered. I'll need to create a translation rule Small-Office VolP Example that replaces anything that could appear as a source calling number and replace it with my username for one of my SIP trunks: R1-PBX(config)#voice translation-rule 1 R1-PBX(cfg-translation-rule)# rule 1 /^.*/ /608222222/ Another problem involves the fact that I've configured the system to require users to dial 9 before calling an external number. Since any call that even makes it to the SIP trunk will have required the caller to dial 9, we'll need to strip those leading 9s from outgoing calls: R1-PBX(config)#voice translation-rule 2 R1-PBX(cfg-translation-rule)# rule 1 /^9/ // Next, we'll need to create a new translation profile that includes these two rules: R1-PBX(config)#voice translation-profile SIP-Outbound R1-PBX(cfg-translation-profile)# translate calling 1 R1-PBX(cfg-translation-profi