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See if we can configure sipp to use separate SIP 'from' for each new call #60

@atsakiridis

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@atsakiridis

That would greatly simplify debugging load testing failures, since we'll have a direct connection between the sipp call inside Restcomm server.log and the Restcomm call inside web console. Would be nice if we could use the sipp auto-generated call-id.

Right now it always uses 'sipp' and to figure out the relationship with webrtc call you need to combine timestamps and other things that are not always easy to find

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