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

[Tavis P]

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)