similar to: Problem with music on hold...

I've followed the docs on how to configure musiconhold. I then added an extension to test it like this: exten => 6601,1,WaitMusicOnHold(30) And when I dial that extension I do hear something for 30 seconds - but it is not what I'd call music... it sounds more like the sound effects from a scifi flick - now and then it sounds like jail cell doors closing and echoing. Kinda cool but

Hi Guys, I've attempted to get this moh-native thing to work with no success. I've reviewed wiki, mantis and e-mail postings and I'm confused. The latest I've read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? I've tried asterisk 1.0.1, 1.0.5, addons 1.0.1, 1.0.4 and also -r stable CVS. I followed the wiki example with

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

Would anyone mind emailing me, or maybe posting somewhere their music on hold .so file? thx -ben

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

A) Modify res_musiconhold.c and the application "WaitMusicOnHold" to accept DTMF breakout B) Create a call queue with a timeout of X and configure the DTMF options properly. Then you can drop callers into this queue and effect a music on hold for X seconds and allow DTMF breakout with no C code. -----Original Message----- From: asterisk@billheckel.com [mailto:asterisk@billheckel.com]

Say there! This is a new Asterisk install on Fedora; I downloaded and installed Asterisk 1.2.1 and it runs fine. I downloaded the 20051217 version of chan_sccp, compiled and it installs ok. When I start Asterisk, it dies on startup at the activation/recognition of the chan_sccp module... A snippit from the startup: Asterisk Dynamic Loader Starting: [res_musiconhold.so] => (Music On Hold

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten

My new asterisk install seems to be running fine - including playing all prompts etc without error. However, when placing someone on hold they here choppy music (first second or so) then quiet. I see the errors below. What is causing this? (Note that I am running AsteriskWin32). Thanks, Mike May 17 17:26:02 VERBOSE[3708]: -- Executing Answer("SIP/2434-263c", "") in

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Trying to test the music on hold function and can't seem to get it to work. If anyone has it running, could you give me a clue? (I have googled and found lots of questions, but no real suggestions.) I downloaded and installed the mpg123 package. From the RH9 console I can start the executable and hear the music via the speakers. The executable is located in /usr/bin. (That works!) I set the

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

Hello, I have upgraded to 1.0.4 version of asterisk. After that asterisk crash every time On receiving an call from iax2 trunk to musiconhold application. SIP calls to MusicOnHold is however working. I already upgraded to 1.0.5, but the problem still Remainig. Any idea ? Iax2 : call proceding : Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching 'WaitMusicOnHold'

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with