similar to: courtesy message calling mobile phones

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

Calling any unreachable mobile from Fastweb or Telecom PSTN I don't get any courtesy message. The same happens when calling an inexistent number. I'm configuring two PBX's, connected to two different phone lines, both behave this way. Perhaps there's some missing zapata parameter? Regards, _fangi_ > Well, > > it's funny because here, now (Italy; Telecom Italia PSTN

Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one

Hi. I'm trying to get a Beronet QuadBRI card work with bristuff drivers. Though qozap module loads right, all card spans are in deactivated status. I'm quite sure my configuration is correct and using a single BRI card instead of the quadBRI the status is active and I can place and receive calls. On Beronet installation manual I read that Beronet and Junghanns cards are identical in

Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

So, I have submitted my configurations as public samples, and I should have expected this situation to arise. I changed all the relevant "private" configuration data in my samples to obfuscate or alter IP addresses, passwords, etc. However, I left my email address in voicemail.conf... Let me tell you, it took THREE messages sent by a distinctly British-sounding gentleman leaving

Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when

Kerry. I would like to get a copy of this document which you speak of. I have had generic trouble in getting livice properly installed and think that would do the trick... ----- Original Message ----- From: "Kerry Cox" <kerry.cox@ksl.com> To: <icecast@xiph.org> Sent: Thursday, November 07, 2002 10:22 AM Subject: Re: [icecast] icecast + liveice won't play nicely

Hi all: I am managing a radio station that has 7 streams. I am using icecast, ices and ezstream on the streams and overall it is working pretty well. One of the streams is an old time radio stream and I use ezstream to run the prerecorded shows. The streams are 128K mp3 streams. Also this station is running on a VPS using Ubuntu 14.04. My question: Is there a package or some sort of

Hi all, I'm sending this to both lists because I want to know what both sides of the conversation think about this. For those of you on the icecast list, can I have a recommendation on the right tool to broadcast with? If you like, I'll send you my prerecorded shows, just tell me what the best way to send those is. For the audacity list now, would it be possible to develop a broadcasting

Hey all, i was just wondering if its possible to play a prerecorded message on a bridged channel. For example : someone calls in on asterisk and is connected through to an outgoing phonenumber, the person picking up on this outgoing call would first hear something like "this call is coming from the asterisk platform". (No problem if the caller on the incoming line hears this too) I

hi, have to host a web-shop which has got the possibility to stream mp3's on demand. those mp3's are prerecorded. is there a possibility to stream local files to the users directly or do i have to use an encoder ?? thanx Juergen --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message

A friend asked me for help to auto-dial and play a prerecorded message for a political campaign. I've briefly googled auto dialer scripts but haven't seen one that really stands out. Are there any free or cheap auto dial solutions that you nice folks recommend? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

At 10:35 AM 7/11/2002 -0500, you wrote: >im resetting up a server for a college radio station and for the new >box we got a decent processor and maxed out the RAM so in addition to >the 2 live encoded streams i want to add 2 prerecorded stream, 1 with >bands that have played live on the air and one of public affairs show, >and i was wondering if darkice and shout will play

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

2012/6/21 Andr?s Gonz?lez <acandido at hi-iberia.es> > ** > On 20/06/12 15:01, Marc wrote: > > Hello List, > > > Hello Marc, > > > as an long time macintosh user , musican/producer/programmer , i am very > upset that another great technology (DSS ) vanished because of http > streaming so i turned my interest towards icecast, whitch seems an >

Hi, I'd like to broadcast the same source at multiple bitrate. Is it possible ? Have you got some clues ? Regards, Pascal <p><p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word 'unsubscribe'