[Asterisk-Users] Problems with MeetMe accepting conference PIN

Dean Thompson Dean.Thompson at learningseat.com
Thu Jan 6 07:01:04 MST 2005


Hi,

I know this question may have been asked before (although the archives 
don't seem to suggest it), but has anyone had any problems with Asterisk 
accepting a PIN number for a conference room.

At this point in time I have established the conference definition in 
the meetme.conf file as well as specifying the appropriate lines in the 
extensions.conf file.

meetme.conf file:
conf => 1234,1574

extensions.conf file:
exten => 1234,1,MeetMe(5678) OR exten => 1234,1,MeetMe,5678

In both cases I can get the SIP client "SJPhone" to actually dial the 
conference number and I hear the operator say, enter the pin number 
followed by a "pound sign".

After entering in the PIN number and entering in the HASH button, the 
system waits for a couple of seconds and then tells me that I have 
entered an invalid PIN number for the conference, please try again.

I have read articles on the web which say that you need a proper RTC 
clocking source for "MeetMe" to work and I have tried both the ztdummy 
module and the "zaprtc-for-2.6" module, but neither seem to be able to 
solve the issue.

At this point in time this is Asterisk CVS version (non-stable) as of 
4/1/05 running on a Fedora Core 3 installation without any ZapTel 
hardware cards.  I have recompiled the kernel and am using 2.6.9 and I 
have both the ztdummy module, zaptel, zaprtc and other modules loaded. 
Asterisk will process normal SIP calls without any problem and allow me 
to leave VoiceMail at the other end, but it refuses to accept the PIN 
number to a conference.

Has anyone had issues similar to this and be able to point me in the 
right direction.  I have also tried using the Asterisk Live CD as well 
and had the same problem regarding the use of conference rooms.

I was beginning to think that SJPhone may well be the problem and that 
it isn't sending the correct tone for the "pound" sign, but I was 
interested to know whether anyone else had this issue when they were 
setting up their Asterisk PBX.

Looking forward to your replies,

Dean Thompson




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