Janine Graves | I don't know how to automate the testing, but I use an XLite 3.0 sip softphone during my training classes and it's always worked really well. The XLite soft phone just needs to be installed and configured to call into the IP address of the gateway on default port 5060 (bypass UCM, don't need CUBE). (Note, there's also a way to configure XLite for work with UCM. I haven't done this, but I know people who have). You can configure XLite with a User Name to match the ANI you want. On the gateway, set up the dial peers to allow sip: dial-peer voice 4075 voip destination-pattern 4075 session protocol sipv2 session target ipv4:10.1.78.75 dtmf-relay rtp-nte codec g711ulaw no vad There also seems to be some sip configuration: ! voice service voip no ip address trusted authenticate allow-connections h323 to h323 allow-connections sip to sip signaling forward unconditional sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 min-se 360 session-expires 360 header-passing ! |
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