similar to: Is this application possible with Asterisk?

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

Hi All, The issue appearing at the random for confbridge module i.e. in some cases if a participant joins the confbridge, he/she unable to hear others which make him/her to hangup the call and redial the bridge again. By joining the bridge second time, participant able to hear the other participants. Any ideas which may causing this issue? As Asterisk version I'm using is 11.2.1. Is it a

Hello guys, I'm using asterisk 11. i'm using Chanspy in a local channel to playBack a file to a specific channel. [playsound] exten => do_playback,1,Answer() same => n,Wait(1) same => n,Playback(${Pv_WhatToPlay}) same => n,Hangup() exten => do_chanspy,1,Answer() same => n,ChanSpy(${Pv_WhoHear},qXBwW) same => n,Hangup() just basically

You're right Steve, sorry for that. So Hi again guys. I need a little help here.I'm using asterisk 11. i'm using Chanspy in a local channel to playBack a file to a specific channel. [playsound] exten => do_playback,1,Answer() same => n,Wait(1) same => n,Playback(${Pv_WhatToPlay}) same => n,Hangup() exten => do_chanspy,1,Answer() same =>

On 12/20/2014 03:22 PM, sean darcy wrote: > On 12/19/2014 09:42 AM, Rusty Newton wrote: >> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >>> I've got a confbridge set up which works if dialed locally: >>> >>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>> --

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised by > Asterisk? I had a bunch of soft phones that I had to change to using ?sip > info? for the DTMF signalling as the RFC signalling was not always being > recognised. This would cause transfers to appear as if the user had not > dialled any digits. > > >

Dear all, i'm using Chanspy to dynamically play a sound file on a specific channel. It works the caller and the callee can hear the file playing during their conversation. However, i'm also using Mixmonitor to record the call. The thing is, in the resulting wav i can of course hear the conversation, but not the sound which was whispering. Anyone knows how to let the whisper being

Hi all, I succeed in injecting audio into one channel by mean of ChanSpy, but this audio cannot be listened correctly. I am using softphones for Android and iPhone and there are so many cuts so that they cannot understand what is said in the audio file. Is this because the total RTP bandwidth is too much?. Which codec and format should I use for the files played with Playback application?

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When

Hi, I recently installed asterisk 1.0.9. Now I want to use the chanspy applicion. Patching and recompiling is working ok. Loading the application into asterisk is now problem either. Now I have the following line in extensions.conf: exten => 556,1,ChanSpy(-b|scan|Agent) When an agent is busy I dial 556 using a SIP Phone. The chanspy application announces that I'm spying on

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

I had a similiar problem, but with calls dropping after 30 sec. It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so when I sent the name of the callee to the caller (as some sort of "centralized phonebook function") it caused calls to be dropped as android refused to reply on the packets or sent rejections back. Check if you have some equipment on the line

The Asterisk Development Team has announced the release of Asterisk 12.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 12.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all