asterisk-user
2006-Sep-29 10:10 UTC
[asterisk-users] [Fwd: [Fwd: [Fwd: asterisk-users Digest, Vol 26, Issue 166]]]
I tried by adding "answer()" to the dial plan but the problem still exists. I am not sure if I am doing this right. Attached is the log file from asterisk while making the call to the conf bridge after adding "answer()" Could you please let me know if you find anything out of this log file? thanks for the help. -------- Original Message -------- Subject: asterisk-users Digest, Vol 26, Issue 166 Date: Thu, 28 Sep 2006 07:42:43 -0700 (MST) From: asterisk-users-request@lists.digium.com Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Message: 19 Date: Thu, 28 Sep 2006 10:30:25 -0400 From: "BJ Weschke" <bweschke@gmail.com> Subject: Re: [asterisk-users] unable to call AT&T audio conference bridge To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <79cf6330609280730y619006a6mab5194b394dd040b@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 9/28/06, asterisk-user <myacc@roundbox.com> 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 asterisk@home <mailto:asterisk@home> did not have > this issue and I even switched back to asterisk@home > <mailto:asterisk@home> (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/ ------------------------------ -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk_debug.zip Type: application/zip Size: 1740 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060929/2021613d/asterisk_debug.zip