asterisk users - Nov 2005 - Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005)

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

tavis

> 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.