On 9/28/06, asterisk-user <[EMAIL PROTECTED]> wrote:
Hello,
I have a problem with asterisk and trying to see if someone can help me
fix the issue...
Problem:
I couldn't join AT&T's Tele Conference bridge directly without their
customer service interaction.
Instead of getting the automated prompts to join the conference, it
takes me to the customer support and then I got to give them the bridge
number and pincode to add me into the conference call.
The reason given by AT&T was that their conference system is unable to
identify our tone.
This happens only with AT&T conference bridges... not sure what the
problem is.
This problem started after I installed trixbox on a new hardware.
Previous setup with [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> did not have
this issue and I even switched back to [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> (a different box) and called the same conf
bridge... that worked fine.
I am running trixbox with the following versions:
asterisk - 1.2.9.1
zaptel - 1.2.8
libpri - 1.2.3-1.349
using zap over a 8 channel pri
Thanks in advance.
AT&T's IVR to collect the passcode is coming through as "early media"
and since you haven't signaled to the phones that the phone is
"answered" they're probably not letting you send DTMF through the
bridge that isn't technically supposed to be there yet.
Put an Answer() in your dial plan prior to sending the call out to
the Dial() application to reach the bridge for these types of calls
and this generally fixes your problems caused by someone else not
signaling correctly.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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