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Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

Rather than using libtool to build its sets, the ao.m4 in beta3 is hard coding '-ldl': AO_LIBS="$AO_LIBS -lao -ldl" Not all systems have a dl library. For now, I can obviously just remove it and keep working on stuff, but what would be a suitable patch that would get accepted to be committed? Should I just check the current build target and set AO_LIBS differently on my

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