similar to: Play music on hold while waiting for DTMF?

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]

Hi all, I am trying to setup music on hold. In extensions.conf did put the following: exten => 6601,1,WaitMusicOnHold(30) Using a sip phone, x-lite, after I dialed 6601 I get the following: -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend) -- Executing WaitMusicOnHold("SIP/666-f408", "30") in new stack Jun 24 22:35:07 WARNING[458766]:

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

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

Hi, all. I'm running the Asterisk 1.0.7 package that comes with Debian Sarge. I'm new to Asterisk, but so far almost everything is going well. However, I'm having a Hell of a time with Music On Hold. I've got an extension set up to test Music On Hold, as follows: exten => 3000,1,WaitMusicOnHold(300) I've tried the following in musiconhold.conf: default =>

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 >

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

I'm using the default mp3 files that ship with Asterisk: fpm-calm-river.mp3 fpm-world-mix.mp3 If they were variable bit rate, I think I would see a warning about 'varibel' or similar...is anyone else able to get these files to work? -Mike- ________________________________ From: Sander [mailto:crombeen@rommelweb.nl] Sent: Tuesday, May 17, 2005 6:34 PM To: 'Asterisk

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

Hi everyone, I've just finished the 'SWait' app for *. SWait = Super Wait :) Syntax: SWait([timeout][cim]) 'timeout' is the number of seconds to wait. Defaults to 'ResponseTimeout'. 'c' is for continue. This changes the default behavior of the app from performing a 'Goto(t,1)' on timeout, to a 'Goto(pri+1)' on timeout, ie. continue to

Josh, As he is playing different streams I think he wants to give the users the option to select another stream. On Mon, May 9, 2016 at 9:58 AM, Joshua Colp <jcolp at digium.com> wrote: > Jonathan H wrote: > >> I'd like multiple people to be able to dial in and listen to various >> live radio streams. >> >> I was told that the correct resource-friendly

I have only been able to confirm this issue with my IAXy as it is the only ATA I have. I am running 1.2.5 stable. Example: User @ exten 100 (iaxy) receives call from PSTN (call 1). While on the call, another user from PSTN (call 2) calls 100 (iaxy) sending a call waiting beep to the iaxy. User 100 answers the call from PSTN call 2 putting PSTN call 1 on hold with music. While call 1 is on

I have a customer who wants their Asterisk system to play an announcement to the caller while a DTMF string is being sent from the caller's phone. As far as I am aware, this isn't possible: it's only possible to detect DTMF if the first incoming digit interrupts the announcement. Is that correct? Is there any way to achieve what the customer is asking for, or is it an unrealistic

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to set up a Linux-based dialin server on our company network. I'd like to have PPP authenticate using winbindd, if possible. I feel like I've almost gotten it to work, but I can't quite get there. Files: /etc/pam.d/ppp: #%PAM-1.0 auth required pam_nologin.so auth sufficient /lib/security/pam_winbind.so account required

[answeringsvc] exten => 0,1,Wait,1 exten => 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r) exten => 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr) exten => 0,103,Goto(0,3) exten => 0,104,Goto(0,3) This should call 713-555-1212. If there are no ZAP lines available it should kick back around and play music on hold until a zap line is available, correct? I'd like the

I would like to do the following: Dial an extension in Asterisk The extension runs an application which dials a number (like a hybrid of DIAL and READ). The dialed number is a box which does nothing but play DTMF tones then hangs up The first box captures the DTMF tones to a variable Dial exits Dialplan continues with the variable value available. Is this possible? Mik

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