similar to: add record and playback sound feature

I downloaded recordPad and recorded a wav file and tried playback on asterisk got the same error as before -- WARNING [1225991360] Format.wav.c:132 check_header:unexpected header size 18-- when I recorded in gsm format on my laptop asterisk did playback well I used sox to resample the recorded wav file on the asterisk machine into wav again and asterisk playback worked well. The sound

I want to have something for the kids to play with which just records until silence is detected, plays back what was recorded, then repeats. They are having fun with Echo() at the moment :) I have mocked something up with: exten => *93,1,Answer exten => *93,n,Record(/tmp/echo:alaw|1) exten => *93,n,Playback(/tmp/echo) exten => *93,n,Goto(2) But it has the shortcomings that a beep is

Hello, >> > I can talk to other SIP phones and via ISDN to the outside, but I >> >don't hear playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played, >> >but I hear nothing at all. >> > >> > Here is as simple example: >> > >> > [monkeys] >> > exten =>

I'm having an issue with an asterisk install that anything recorded (in .gsm format) and all of the pre-recorded .gsm files are choppy. All calls into the asterisk box is fine and any voice mails left in a box are fine as well. It's just the playback of any recorded message and any of the pre-recorded files. Anyone have an idea what might be going on? The only problem is the playback of

I'd like to be able to playback a file to *both* channels in a call as a result of a DTMF feature. Can anyone suggest how I might do this? I thought of using a DYNAMIC_FEATURE to call a macro that starts a dynamic meetme.... but the macro only gets to control the 'caller' or 'callee' :-( Failing that.... I'm trying to provide a simple means of playing back a recorded

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Hi, we are using Asterisk to record and playback. Both services are working well independently but it seems we can't start playback of a file while we are still recording it, even if the file is already in the hard disk. Is it possible to playback while recording the same audio file? Or is there a way to enable it? Regards, Jahnavi. -------------- next part -------------- An HTML attachment

Dear Group, I'm running the following setup; Yoper v2, Kernel 2.6.8.1-7, Wine 20040914 on a Toshiba Satellite Pro 4600. The Toshiba has a Yamaha AC-XG. In addition I have a USB Plantronics DSP100. After some tweaking I got wine to install x-ten lite; I have pasted my config file; [WinMM] ; Wine supports the following sound drivers: ; winearts.drv ; for KDE ; winealsa.drv ; for

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

>On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: >> > Hello, >> > >> > I have a strange situation: >> > >> > I can talk to other SIP phones and via ISDN to the outside, but I >don't hear >> > playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played,

For some applications, storing recorded audio (prompts and caller recordings) as a BLOB in MySQL has advantages. So, once I have the audio in the database, how can I play it? Creating temporary files seems so tacky. Is there another way to playback or background audio either by specifying a URL or from a memory buffer (either C or PHP)? -- Thanks in advance,

New install, Calls are working phone to phone using gsm, ulaw or alaw codec but when try and echo test or voicemail there is no playback. I've tried turning on and off every codec and still no luck. Asterisk says it's playing the sound file but I just don't hear anything. I can't find any reason for this. I've tried the latest tar and CVS with the same result.

Hi, I've installed asterisk from the Ubuntu Dapper packages, and it has been running for several weeks absolutely fine - calls work both in and out (only using SIP on both sides). I've come to setup Voicemail now, but when I dial the extension with VoiceMailMain, I see: -- Executing VoiceMailMain("SIP/pettit-bedroom-9964", "@pettit") in new stack == Auto

Thank you for you quick reply, Jean-Marc. I have just used a human voice to replace the sine wave, but the result is like a broken voice after tens of seconds. Do you have some sample voice in .sw that I can test with? I want to have one set or a few sets of sample voice that work for my understanding and debugging. This is the output from echo_diagnostic.m in my human voice test: Far end to

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

A small patch (attached) to the ogg123 source (dec 15 nightly build from CVS) that implements ranged playback a la: ogg123 -r 12:10-13:00 file.ogg to playback 12m:10s-13m:00s fragment from the file.ogg soundfile. Usage: ogg123 -r hours:minutes:seconds.fraction-hours:minutes:seconds.fraction anything can be pretty much omitted (within reason) [although it does not support hh:mm:ss.hh- ]

Ever since upgrading from 1.2.X to 1.4.2, I'm having trouble with voicemail. When played back, the messages start out okay, but after 10 seconds or so, the playback speed starts to increase and the voice becomes illegible. It seems like some kind of audio timing problem. Phone calls seem okay, in general. I don't have any digium cards, but I am using ztdummy. Ztdummy is loaded

Hi, I would like to implement (with Asterisk 1.6) something like this : - when you answer a call and a fax is detected by Asterisk, the other party is forwarded to ReceiveFax application and a pre-recorded message "You are receiving an incoming fax" is played to the receiving party. Using fax standard extension, I can handle the first part (forwarding caller to ReceiveFax app) but I

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone