Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right:
sip:[EMAIL PROTECTED] Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is asterisk-1.0_RC1. No NAT. The phones I've tried so far are as follows. ** Linphone: Check out my SRPM at http://www.bigu.org/SRPMS/ Sound is fine. Doesn't seem to pick up anything from the microphone, though. ** kphone: Check out my SRPM at http://www.bigu.org/SRPMS/ Sound is fine. Picks up sound from the microphone, but the echo-test repeats it back after passing it through a Mr-Roboto filter. ** tkPhone: Sound is fine. Doesn't seem to be reading from the mic, no traffic going over the network after the 'demo-echotest' recording finishes. The following errors are continuously repeated from tkPhone: sent 63426 (3),received 30228 (3);read 615040 write 293120 need 608000 jitter 38!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!! Data won't fit within the current RTP packet size ** SJphone: Last night: sound worked fine. Actually sent sound from the mic, which came back after about a 5-second delay, but which sounded quite good. Today: establishes a connection, but absolutely no sound in or out. :P If anyone's interested in my SRPMs, I'd like to know. The asterisk RPM builds the zapata, zaptel, and asterisk sources; you must have your kernel-source rpm installed for it to build the modules against. Let me know if there's something obvious I'm missing. Thanks- John _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users