Rich Adamson wrote: >>I'm running Asterisk 1.2.0b2 (also tried latest CVS HEAD) in my lab and >>i've come across a strange problem. >> >>I've setup an extension to call the meetme application, when i call that >>extension it functions as expected, informing me of my conference number >>and that i'm the only one in the conference however right after join the >>conference some problems start occuring: >> >>1. If i call in with another client (both are SIP based), it does not >>acknowledge the DTMF tones i send to select the conference room, it acts >>like it never received the DTMF (it plays the "please enter the >>conference number followed by the pound key" prompt again) >>I have verified that the tones are being sent properly, and otherwise >>work as expected. (before selecting a conference room) >> >>2. When i hang up the phone Asterisk does not clear the SIP channel in >>use by that phone. >>Before selecting a conference room calls are properly disconnected by >>Asterisk and removed from the "sip show channels" list. >> >>3. After the RTP timeout hits (as configured in sip.conf) it prints a >>message every second that the call has timed out and will be >>disconnected. This continues on forever it seems (12 hours in one case) >>Before selecting a conference room, if left idle (no RTP is sent from >>SIP UAC), the SIP session is properly disconnected/terminated after the >>RTP idle timer hits. >> >>if add the "de" options (dynamic, select an empty conference room) >>the first caller hears the meetme prompts and is put into the first >>conference room, however the second caller hears nothing, looking at the >>debug output on asterisk shows that meetme was called and nothing else >>after that >> >> >>I'm running on linux kernel 2.6.13.4 (vanilla, with grsecurity patches) >>Zaptel drivers were compiled with "make linux26" >>There is a T100P card in the system and the "zaptel" and "wct1xxp" >>modules are loaded >>I've tried using the ztdummy module in place of wct1xxp with the same >>results >>Asterisk and Zaptel were compiled with gcc 3.3.5 on Debian Sarge >> >>submitted bug - http://bugs.digium.com/view.php?id=5578 >> >> > >That's odd. I just checked our meetme using two C7960's and an external >Zap (pstn) call, and all worked as expected. Using cvs-head from early >morning Nov 1 on fc3 with analog TDM04 card. > > > >
Were you using the SIP voice software on those 7960s? I was using one Cisco 7960 (7.3) and one Sipura SPA2100 (3.1.3) _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
