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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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I will try to comment on some of the things you brought up starting from the end. Please correct me if I misunderstood something.
SIP forking is supported by PBXes in the form of Ring Groups and Queues. An incoming call sent to a Ring Group or Queue allows any extension in it to pickup the call.
PBXes can act both as a SIP Proxy as well as a B2BUA depending on a few parameters. When calling from extension to extension, if both of them support re-invites (audio bypass set to Yes), then the remote IP can be seen on a trace, since the RTP packets bypass the proxy completely.
The requirement for the above scenario, is for the other party to pickup the phone. If they don't and the call ends up in VM, you are out of luck. Once you have their IP though, you can make a call directly to their device.
The question is, why you would go through all that trouble, if they already have a registered extension on PBXes. If their extension is no longer registered, chances are their UA is offline so even a direct call to it will not complete.
Having said that, I can see the value of displaying the IP of every SIP extension for troubleshooting purposes. We can avoid setting up a packet trace just to discover the IP of a UA, so we can ping it to determine the current packet loss. Or asking the user at the other side for their IP.
Pascal can you please comment on the feasibility of this request? It can make our lives easier, while the information is already available.
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22.12.2006 05:47 |
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