Thanks for the help here… Let me know if you need any further information.
Do you get this problem on Asterisk 1.8.4? Which version are you actually using?
1.8.4 Built from source.
What does the verbose CLI show?
Exactly the same as when a normal pin code is punched in, no errors/warnings. (it is worth noting that if I use a normal pin code (as opposed to admin) from the same phone it works fine. )
Which suggestions from your Google searches did you try?
I haven’t tried anything as I was unable to find anyone else experiancing this problem.
What is your timing source for meetme?
dahdi, I have checked and this is functioning correctly.
What “technology” (Asterisk terms) are you using for the participants?
Sorry, I dont really understand this question.
The following assume that you are using SIP:
What do the SDP exchanges show?
Nothing out of the ordinary… I do beleive this is a conference issue rather than a sip/firewall/nat one as other calls work fine.
Do you have NAT in your network?
What is in sip.conf to account for the NAT?
It is configured to account for NAT yes, worth noting that other calls to sip clients, landlines, mobiles etc all work correctly.
Do you have stateful firewalls?
Yes - it is worth noting I have turned off the firewalls (put machines on DMZ) to test this and still get the issue.
Are you using a loud speaking phone?
If so, what is its VOX sensitivity set to?
Are the other people using loud speaking phones?
What is their anti-VOX sensitivity?
Why didn’t you mention one way audio in the subject?
Forgot, sorry! corrected this now.