similar to: whisper time remaining

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Greetings, I have tried numerous ways to set the LIMIT_PLAYAUDIO_CALLER and LIMIT_PLAYAUDIO_CALLEE variables with no success. The default parameters never change. Has anyone had success changing the defaults? If so, how did you do it? Thanks, vcomp -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

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

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

Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached) same => n,Dial(Local/s at root/n,3,L(3540000:60000)) same => n,Hangup() [root] exten

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

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Asterisk 14.1 Here's a bit of test dialplan, which works as expected and simulates exactly what I'm doing at the top of my large dialplan... [dial-pre-test] exten => s,1,NoOp() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) same => n,Dial(Local/s at dial-test,3,L(3540000:60000)) same => n,Hangup() [dial-test]

Can someone tell me how, if it has been done. What I an looking for is the ability to have a assistant or other authorized person 'whisper' in my ear duing a conversation. The party on the remote end would not hear anything. I can do a redirect into a meetme room with the person on the remote end in a speak only (t) and the other two as regular conferancees. After short time, break down

Dear all: I want to setup a "whispering" (or "coach") meetme room. the goal is make a channel with the capability of barge in a Agent call with a customer, but only the operator can ear the voice of the supervisor. This with the purpose agent monitoring. Anybody out there can set-up this function? The only parameters in Meetme app ('m' and 't') don't

Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.....do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can?t be upgraded (policy

Hi, Has anyone tried using these flags for the Dial command? I set it to "no" but both parties can still hear the (beep) warning sound. - Wei

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?

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? :^)