similar to: Injecting a sound file into a bridged call

Hello everyone, I'm considering getting me a quad-core Dell PowerEdge 860 to run Asterisk. Anyone else using this model? Any tales of woe and sorrow I should know about? Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and giving that a try. Same question with that one - any quirks I should be aware of? Girts -------------- next part -------------- An HTML

Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a sound file playing a message. The problem is that for some reason the MixMonitor does not record whatever

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well as a lack of spy/whispering commands available via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

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

http://www.youtube.com/watch?v=tDacjrSCeq4 I wonder if the inverse is true. If I whisper soothing words of encouragement at my JBODs, will I get more IOPS with reduced latency? :^)

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simply playing around. My background however is with Cisco CallManager, Cisco IPCCX for call centers as

Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/21cd5c04/attachment.htm

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

Hi, I''ve used .save to save the attributes, but on my table there is an autoincrementing id (not the primary key) that doesn''t need to be inserted... but save try to INSERT it anyway... is there a way to avoid save from acting this way or to whisper him to be more polite this time? ;) Thanks, Enrico -- "The only thing necessary for the triumph of evil is for good men to do

Hi all, I just noticed this: https://sourceforge.net/p/flac/feature-requests/91/ a request for support of 32 bit audio data. The request has been around since 2008. Had two inial impressions: * Would adding this break brackwards compatibility too badly? Obviously decoding of 32 bit encoded data would not work with older versions of flac. * This is nuts. 24 bits has a dynamic range of

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

After failing to install on two different machines that were running Windows, I realized that one machine had a bad hard-drive, and the other bad memory. So I've collected the good parts into one machine and installed CentOS with SME. This was the fastest install of Linux I have every seen, 15 minutes done, configured, and running, web and email server. It is a very secure config, no GUI

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

Hello, We're having a problem in Dovecot 1.1.8 with a failed assertion on certain mbox format mailboxes. It happens both with deliver when it attempts to delier to the mailbox, and with IMAP connections for the affected box (though I'm not sure what they're doing at the time). Mar 3 12:55:26 <snip> dovecot: Panic: IMAP(<snip>): file istream.c: line 81 (i_stream_read):

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com