search for: rerecord

Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD

...)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten => confmenu,1,Background(to-listen-to-it&press-1&to-rerecord-it&press-star|m||macro-systemrecording) exten => confmenu,n,Read(RECRESULT||1|||4) exten => confmenu,n,GotoIf($["x${RECRESULT}"="x*"]?dorecord,1) exten => confmenu,n,GotoIf($["x${RECRESULT}"="x1"]?docheck,1) exten => confmenu,n,Goto(1) exten =...

...AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so good, escape character) * fast forward/rewind the recording as it is so far (!) * overdub (i.e. rerecord) starting from a desired point in the existing sound file (!!!) So, it's pretty clear that AGI's "RECORD FILE" function as it stands is not going to cut it, so I'm assuming that do anything like this means writing a new custom app_something.c, does that sound right? Anybody h...

Hi, I'm testing Asterisk installation on my Slackware Linux Box, I am making calls from kphone. I've noticed that standard sample voice files attached to * distro (*.gdm files) havn't good quality. What can be done to improve it? I'm using alsa drivers. BR, Corvin

Hi! If a user has recorded a new voicemail message (e.g. unavailable message) then it is prompted with 3 choices. 1. accept recording 2. listen to the recorded message 3. rerecord the message Isn't it possible to cancel the recording? thanks klaus

I want to rerecord the "1" "2" "3" ... "0" sounds, but not overwrite the defaults. So, I've recorded them into a custom directory /var/lib/asterisk/sounds/custom I was hoping to be able to do the following: exten => foo,1,Set(CHANNEL(language)=custom) exten => foo...

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

Just got a call from Ebay's unwired buyer and "The Voice" is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). Thanks, Steve

...) (NEW) (see state D) [0] to reach operator (as defined in extensions.conf) (no message is left in vm) [#] if listening to greeting, skips rest of greeting and jumps to recording message [#] - if recording a message ends recording [*] - Cancel message (erase recording) (gets message please rerecord your message) [1] - Accept recording [2] - Review (listen to) recording [3] - Erase and re-record [0] - Reach operator (as defined in extensions.conf) (no message is left) State B - in vm as a vm box owner. System has associated vm box (maybe based on caller ID). [*] - cancel...

...; if not listed, sending messages from inside voicemail will not be ; permitted ; callback=fromvm ; Context to call back from ; if not listed, calling the sender back will not be permitted ; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default ; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to ; reach an operator [OFF by default] ; envelope=no ; Turn on/off envelope playback before message playback. [ON by defaul...

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

...b\tab\tab ; if not listed, sending messages from inside voicemail will not be \par \tab\tab\tab ; permitted\par ; callback=fromvm \tab ; Context to call back from \par \tab\tab\tab ; if not listed, calling the sender back will not be permitted\par ; review=yes \tab\tab ; Allow sender to review/rerecord their message before saving it [OFF by default\par ; operator=yes \tab\tab ; Allow sender to hit 0 before/after/during leaving a voicemail to \par \tab\tab\tab ; reach an operator [OFF by default]\par ; envelope=no \tab\tab ; Turn on/off envelope playback before message playback. [ON by defau...

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal

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