similar to: AGI() or System() - Play audio in the background

Just run ‘core show calls’ as a command from the AMI, and parse the results. I don’t think there is an equivalent pure AMI command. From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent: Sunday, June 14, 2020 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re:

Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I

Hi, Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I did not get it working. This is my musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [general] [default] mode=custum

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you

I am having trouble getting streaming MOH to work. As far as I can tell I have everything configured properly but there is only silence. Your help is appreciated. I am running Asterisk 1.8.11-cert10 with mpg123 1.12.1 to play the stream (I have tried madplay, and mpg321, and I compiled streamplayer as well with the same results). I started by finding a working stream and tested this from the shell

Way back in the mists of time, I built my asterisk installation with SNMP support. It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that moment in time. That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less

Wow! I've been *-ing for about 6 years and had literally no idea about that! I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit hacky) dialplan. That's making me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards,

MP3s have to use constant bitrate not variable bit rate. Look in the documentation for mpg123. -----Original Message----- From: Jozeph Brasil [mailto:jozeph@globalmedia.com.br] Sent: Saturday, July 24, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: RES: [Asterisk-Users] Play CD! I do that. But when I play MusicOnHold the music is played slowly! I don?t know why... but is how

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

hi All, I am using 11.25.2 and musing on hold. CentOS 7.4 I am trying to setup a MusicOnHold() streaming audio. I have one machine I tried this on and it worked. This machine is on the internet. I have another machine behind a firewall that does not play. Both machines installed mpg123. I have a windows machine behind the firewall that plays the audio stream so firewall is not the issue. I

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

Hi all, A long time ago I built an Asterisk system that plays IceCast streams via moh. extensions.conf: Exten => moh,1,Set(SIP_CODEC=ulaw) Exten => moh,2,Answer Exten => moh,3,MusicONHold(test_new) Exten => moh,4,Hangup musiconhold.conf ; test_new [test_new] mode=custom application=/etc/mystreams/test_new.sh test_new.sh #!/bin/bash wget -q -T 120 -O -

Thank you... but "just update the database" - hmm, what database? Did you mean ARI? I still can't find the command! The asterisk wiki is somewhat, um... spread around! On Sat, 13 Jun 2020 at 16:56, Steve Edwards <asterisk.org at sedwards.com> wrote: > > On Sat, 13 Jun 2020, Jonathan H wrote: > > > I need to ensure that a MusicOnHold stream is only running when

I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me this. Any ideas?!? In case it helps and you're wondering why... I need to ensure

Hi, i am having problem on playing asterisk sound file on my new installed asterisk.. i have the following extension , if i call from any SIP / IAX phone playback or voicemail doesnt play anything .... but when i dial 102, I hear the MP3 music .. exten => 99,1,Answer() exten => 99,2,Playback(prepaid-welcome) exten => 99,3,Hangup() exten => 101,1,VoiceMailMain() exten =>

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

First off, let me first say that this is not a one-way audio problem. Sometimes I can get 'her' to speak to me, other times I can't for a long time. I'm just using a very simple system to dump people into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf

I've been working on setting up a few queues and agents for use in our upcoming call center. Agents can log in and customers can call in and be routed properly, but I've run into one strange issue. When an agent dials 98 to login the "agent-incorrect" file is played, then the agent can login as normal. Searching the archives I found this issue brought up in Feb '04 :

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the