Hello all, I've build a traffic generator around sipXtapi (win32), which we're using here to test our (IP)PBX'es. By using testscrips, an n-amount of SIP UA's are created and registered on which different call patterns are executed. It works quite well, but I'm running into two questions/problems for which I want to consult you:
1. No media prefered I want to be able to disable/bypass the media. Currently sipXtapi automatically 'connects' to the PC's audio device and is using this. Preferly I don't want sipXtapi to use any media (no sending of RTP and just dropping received RTP). Are there compiler switches (preprocessor defines?) or is there a runtime method call? 2. Problem with transfer Imagine next scenario (all UA's involved are created within sipXtapi): - UA-1 makes call to UA-2 - UA-2 answers this call => both connected state - UA-2 puts UA-1 on hold - UA-2 makes (consulting) call to UA-3 - UA-3 answers the call (or does not answer -> makes no sense) - UA-2 now executes the sipxCallTransfer to transfer UA-1 to UA-3 ==> I'm now getting an *CALLSTATE_CAUSE_RESOURCE_LIMIT *error?!?! Resources cannot be the problem, because I can have multiple UA's around (>10) and in above example I only had 3 UA's active. Anybody a clue what is going wrong here? With regards, EiSl
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