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

[Tavis P]

Tavis P wrote:

>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.
>>
>>
>> 
>>
>>    
>>


It seems that this issue is related to my use of the wct1xxp module and
a T100P T1 card.

After removing the wct1xxp module and loading the ztdummy module in its
place the conference bridge works as expected, Asterisk removes
(properly) terminated sessions and times out idle sessions.

Using the ztdummy module in place of wct1xxp shows a noticable drop in
audio quality (blips and phasing)