identifying sip uri call & currentCall->network values

How do I determine that a call is arriving on the SIP URI? currentCall->calledID is the same for SIP and Skype calls, i.e., the complete SIP URI / Skype number is not provided, only the 999xxxxxxx that is common to both SIP and Skype calls. I thought currentCall->network would be useful. The documentation says for voice the options are PSTN or VOIP. However I've noticed a few things: 1) there are undocumented values for currentCall->network, e.g., SIP & SKYPE 2) I've never seen PSTN or VOIP despite having received both types of calls. 3) Instead, SIP, mobile and POTS calls all have the undocumented value SIP and Skype calls have the undocumented value SKYPE. I suppose I could test for ((calledID == 999xxxxxxx && (network != skype)) But it seems like a kludge and relies on the undocumented network values and the shorted SIP/Skype calledID. In addition to an answer to the first question at the top, could we also have updated documentation on currentCall->network? Jim


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