[Asterisk-Users] Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005)

[Rich Adamson]

> >>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)

Our C7960's are running v7.1 (no need to upgrade), and just tested
with a SPA3000 v3.1.7g. Works fine.