similar to: Announcement Transfer with call-limit = 1

Hello List, I want to play an announcement for attended transfer calls. For example, "A" calls "B", "B" answers the call and transfers (attended) to "C" - once transfer is complete "B" should hear an announcement saying "you call has been transferred". Is there any configuration in asterisk to implement this behavior? I have not

Hi anyone, I need to install an application to organize conference in Asterisk, and to this I wanna use webmeetme, but I don't get find a good manual, anyone have or know where I can find a good manual to this application? Thank you very much. Renato -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I wanna record the conversations in a conference call, anyone know how can I do it? I've already configurated a room on meetme.conf but I don't know as I can record the conversations. I'm using SUSE 11 and Asterisk 1.6.2. Thank you so much for help me. Bye ____________________________________________________________________________________ Veja quais s?o os

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Hello List, Please help with the following problem, I have a situation, where I need to play an audio announcement to the caller SIP channel once an attended transfer is successful. The attended transfer is done from client. I can see a transfer event in AMI. I am not using 'T/t' option in dial() command. The transfer is completely on client side using SIP signaling. 1. A calls B 2. B

I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a "this call may be recorded" announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the

Hi all ,I am using Asterisk-1.6 and facing a problem during call transfer.when an agent transfer the call to another agent , there is ?no entry in the queue_log file.is this a problem in Asterisk or i am doing some wrong?? PLEASE HELP........................................... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there a way to do a blind transfer within an asterisk dialplan (like '##')? The reason I need this (I think) rather than a regular Goto() is that I'm trying to do one-touch parking. I can park a call using one-touch parking and then pick it up again, however if I try to re-park the call, it gets lost. I think that is because asterisk thinks I'm still on the park extension.

Hi all does anyone knows if there is a parameter to limit the amount of calls on an ACD Queue? I saw somewhere a reference to a QUEUESIZE paramter but couldn't find anything else. This is to work as a backup plan for very specific emergency situations, for example I have 60 channels available, and usually at midnight only 3 operators are needed to handle a normal flow. But on some

On Apr 3, 2013, at 2:07 AM, Renato Golin <renato.golin at linaro.org> wrote: > On 1 April 2013 22:05, Bill Wendling <wendling at apple.com> wrote: > We would like to support ARM again. > > Hi Bill, > > Glad you asked! ;) > > I'm getting the test-suite bot green (a few minor tweaks and we're good) and that should get us well ahead of what we've

Hi All; Can I do transfer for the call from zap/1 to zap/2 (both are fxs)? All what I need is to add the t argument for the Dail function? And how can I transfer to be in that senario: zap/1 dial a code to transfer for zap/2, once zap/2 answered, then he can talk with zap/1 (where the third party will not hear what they are talking), then if zap/1 hanged up the handset, the call will be between

Call for Participation http://www.hpdc.org/2012/ The organizing committee is delighted to invite you to *HPDC'12*, the /21st International ACM Symposium on High-Performance Parallel and Distributed Computing/, to be held in *Delft, the Netherlands*, which is a historic, picturesque city that is less than one hour away from Amsterdam-Schiphol airport. HPDC <http://www.hpdc.org> is

Call for Participation http://www.hpdc.org/2012/ The organizing committee is delighted to invite you to *HPDC'12*, the /21st International ACM Symposium on High-Performance Parallel and Distributed Computing/, to be held in *Delft, the Netherlands*, which is a historic, picturesque city that is less than one hour away from Amsterdam-Schiphol airport. HPDC <http://www.hpdc.org> is

On 1 April 2013 22:05, Bill Wendling <wendling at apple.com> wrote: > We would like to support ARM again. > Hi Bill, Glad you asked! ;) I'm getting the test-suite bot green (a few minor tweaks and we're good) and that should get us well ahead of what we've ever been on ARM. Though, bootstrapping seem to fails a few check-all tests. I'll look into that as soon as the

With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says "so and so is on the phone for you", I say "ok put him through", she hangs up and I am connected to the caller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

I know this topic has been done before but I cant find an answer for it anywhere. I have Grandstream phones running with my Asterisk server. Is there anyway to do announced transfers with these phones? i.e. A is talking to B, A then presses a key to initiate transfer, calls C, tells C that B is on the phone. Then A hangs up and C is connected to B.

On 9 April 2013 09:48, 陳韋任 (Wei-Ren Chen) <chenwj at iis.sinica.edu.tw> wrote: > > We are currently in the process of switching FreeBSD/ARM over to clang. > We will certainly do testing, but I'm not sure it's worth LLVM shipping > binaries, as clang 3.3 will be the system compiler and clang trunk is > available in ports. > > I think the arm binary is for

On 3 April 2013 21:15, Bill Wendling <wendling at apple.com> wrote: > > > * Green direct check-all (clang-native-arm-cortex-a9 bot, green) > > * Green direct test-suite (clang-native-arm-lnt bot, almost green) > > * Green self-host check-all (no bot yet, some failures, will look into it next) > > All of this, yes. :) Great! This is feasible for the 3.3 time frame.