search for: conferece

I've got one that I haven't been able to solve. Hopefully someone else has had this issue. I'm using the paging script in free pbx, which appears to: Send a sipheader autoanswer, Create a conferece Add the phone to the conference But if the user hits the page extension, all the phones auto answer, and if they hang-up before the phones join the conference I end up with dozens of phones off the hook which never hang up. Anyone else seen this or have a solution? Thanks! --Jeffrey ----------...

I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have

Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk.

Does anyone have an example extensions.conf section for initiating a three-way call? I don't see documentation on the syntax anywhere, and haven't been able to figure it out from the source. Thanks! Joel

hallo, i tried to setup my extentions,conf like this but it never jumps to the busy part (102) asterisk always plays the unavail msg, also when i am connected to another iax channel (conferece room) and no more channel on my client is available. could sombody give me a hint what could be wrong? thanks , alex snd*CLI> -- Accepting AUTHENTICATED call from 81.135.10.114, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2/krath@krath/3", "IA...

...ed" back from 191.21.21.21 > Channel SIP/0041435215309-3c5a was never answered. == Manager 'manager' logged off from 191.21.21.21 I want to create a dump connection between a dump extension to any extension then redirect the channel from the dump extension side to the conferece. but How can i make the dump extension to auto-answer and create a channel when I Originate Call using manager API? Best Ehsan --------------------------------- Sponsored Link $420,000 Mortgage for $1,399/month - Think You Pay Too Much For Your Mortgage? Find Out! -------------- next...

Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact configuration in dialplan and any to edit confugration files please help me . and how can they cut the conference of after concall. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limite...

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press