Re: [asterisk-users] file command with alaw file
From alaw to wav, you can use Asterisk's CLI f file convert /var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav I want to have my record alaw automically be converted to mp3 (or wav) right after finishing recording. How can I do it in dialplan? We are going to connect our asterisk to E1-Euro ISDN. AFASIK, it uses G711 A-law for audio codec. Is it possible for me to save my recorded audio file directly as wav files without calling CLI's file command (it will be converted to mp3 later)? Thanks for all your time and your wisdom Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] file command with alaw file
Hi, How can I convert FROM ALAW file, which generated by asterisk apps (monitor, or record app), to format (wav or mp3) that is playable by music player?? Can Sox do this? I have an asterisk 1.6.2.6 on my CentOS 5.2. I record audio clip by mixmonitor app and use file command to check the alaw output, and here is output: - 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz - How could file command recognize the format as there is no header in the output file? Or Did I probably miss something making asterisk yield incorrect alaw files? Please help, thanks Quyps On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote: Quyps-- I've noticed in general that the ulaw, alaw, gsm, slin files used and generated by asterisk do not have headers (the RIFF stuff), and asterisk is not expecting them. in general they will louse up Asterisk's ability to play the sound. They are just raw data and the extension on the file name (.gsm, or .ulaw, etc) is the only clue to the file format/contents. In general, if you need a sound file of your own making in a certain format, you can convert to most of the formats using sox in linux, but really, the best thing to do is record the source sound file in cd-quality sound WAV format, in 44 khz sampling rate, or higher, and then use sox to convert to 8khz format. Asterisk can do some of this via the file convert CLI command, ( on the asterisk cli, type help file convert). You'd have to judge for yourself if file convert tt-weasels.gsm tt-weasels.ulaw which would convert the 8khz gsm format to 8 khz ulaw, or sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t sw tt-weasels.raw; sox -r 8000 -c 1 -t sw tt-weasels.raw -t ul tt-weasels.ulaw which is the way the Asterisk sounds are produced from the the cd-quality sounds. They would seem a bit equivalent. I wonder if just sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t ul tt-weasels.ulaw would sound any better... you audio engineers out there may have an opinion. I've personally noted that not all linux distributions provide the same version of sox; some distribute sox with an absolute minimum of sound formats built-in. You may have to go out and find all the libraries and roll your own sox. murf On Wed, May 19, 2010 at 10:34 PM, Pham Quy qu...@vega.com.vn wrote: On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: hi all, I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record audio clip with mixmonitor() as alaw file (softphone is also configured with alaw active only). Using file command i can get the following information 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i could get the same information with file command. File command recognized alaw file as JPEG image: 983006584-20100517-123825.alaw: JPEG image data I guess i may miss something when i setup the new on on Centos 5.5, but u dont know what it is. Anyone have idea about this? please help. Thanks in advance. Quyps I did check content of two alaw files (using a hex editor) generated from two aboves cases. For the one fromo CentOS 5.2, beginning bytes look likes : riff1^0.wavefmt@...@...fact.^0.data.^0... and the one from CentOS 5.5 ..RQVTVXEMBAX It seem like the first one have some information about file format, which make our convert tool works correctly, and which the second one doesnt have. Can you explain to me this different, and how can i get the information as the first one? Thanks in advances, Quyps This question have been asked for a while, I really need some help here? Thanks in advance. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] file command with alaw file
On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: hi all, I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record audio clip with mixmonitor() as alaw file (softphone is also configured with alaw active only). Using file command i can get the following information 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i could get the same information with file command. File command recognized alaw file as JPEG image: 983006584-20100517-123825.alaw: JPEG image data I guess i may miss something when i setup the new on on Centos 5.5, but u dont know what it is. Anyone have idea about this? please help. Thanks in advance. Quyps I did check content of two alaw files (using a hex editor) generated from two aboves cases. For the one fromo CentOS 5.2, beginning bytes look likes : riff1^0.wavefmt@...@...fact.^0.data.^0... and the one from CentOS 5.5 ..RQVTVXEMBAX It seem like the first one have some information about file format, which make our convert tool works correctly, and which the second one doesnt have. Can you explain to me this different, and how can i get the information as the first one? Thanks in advances, Quyps This question have been asked for a while, I really need some help here? Thanks in advance. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] file command with alaw file
hi all, I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record audio clip with mixmonitor() as alaw file (softphone is also configured with alaw active only). Using file command i can get the following information 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i could get the same information with file command. File command recognized alaw file as JPEG image: 983006584-20100517-123825.alaw: JPEG image data I guess i may miss something when i setup the new on on Centos 5.5, but u dont know what it is. Anyone have idea about this? please help. Thanks in advance. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] file command with alaw file
On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: hi all, I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record audio clip with mixmonitor() as alaw file (softphone is also configured with alaw active only). Using file command i can get the following information 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5 #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i could get the same information with file command. File command recognized alaw file as JPEG image: 983006584-20100517-123825.alaw: JPEG image data I guess i may miss something when i setup the new on on Centos 5.5, but u dont know what it is. Anyone have idea about this? please help. Thanks in advance. Quyps I did check content of two alaw files (using a hex editor) generated from two aboves cases. For the one fromo CentOS 5.2, beginning bytes look likes : riff1^0.wavefmt@...@...fact.^0.data.^0... and the one from CentOS 5.5 ..RQVTVXEMBAX It seem like the first one have some information about file format, which make our convert tool works correctly, and which the second one doesnt have. Can you explain to me this different, and how can i get the information as the first one? Thanks in advances, Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
On Wed, 2010-05-12 at 22:10 -0700, Steve Edwards wrote:
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
The problem is...
You have no clue[s] :)
First off, the module should be cdr_addon_mysql.so, not cdr_addon_mysql.o.
If you don't have the so in /usr/lib/asterisk/modules/ something is
wrong with your build.
Try something like this:
sudo -u whatever-user-runs-asterisk-on-your-system\
/usr/sbin/asterisk -c -d -d -d -f -g -n -v -v -v
Or, you can start Asterisk without loading cdr_addon_mysql.so and then
load it from the Asterisk CLI. It sounds like you are auto-loading modules
so you could add noload=cdr_addon_mysql.so to
/etc/asterisk/modules.conf to get Asterisk running and then load it with
something like load cdr_addon_mysql.so
I'm a 1.2 Luddite so the commands may have changed slightly. Also,
depending on the specifics of your installation, the paths may be
different.
See if this gives you any clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Hi,
Yes, it was a typing mistake, i meant cdr_addon_mysql.so. After manually
loadind the module, it turn out there is an mistake in my cdr_mysql.conf
I fixed it and everything work fine.
Thanks.
Quyps
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[asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
Thanks in advance.
Quyps
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Re: [asterisk-users] Does 'file' command work with asterisk genereted alaw file
On Sun, 2010-04-25 at 23:39 -0500, Tilghman Lesher wrote: On Sunday 25 April 2010 23:22:21 Pham Quy wrote: I record an alaw file by asterisk's record monitor command, and i use linux's file command to check it information. file command recognized the alaw file as DATA, is it correct? The file command works by recognizing certain header data in a file. As the alaw format has no header, that command won't recognize the data as any particular format, so it tells you DATA as its fallback. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org I've just tried to config my softphone (twinkle) with different codecs. i found out that if I remove all the active codecs but G.711 A-law, then the .alaw-output file can be recognized by 'file' command as following: #file 983006584-20100426-142120.alaw 983006584-20100426-142120.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz - but after i changed the active codec to the others, the output is recognize as DATA again. Does the .alaw-output internal codec (or whatever it is) depends on the codec used by softphone? and if i choose different softphone codec, i will get different output file? Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does 'file' command work with asterisk genereted alaw file
Hi, I record an alaw file by asterisk's record monitor command, and i use linux's file command to check it information. file command recognized the alaw file as DATA, is it correct? Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk segmentation fault
Hi all,
I have a problem with my asterisk. When i start asterisk, i got the
following
--
/usr/sbin/safe_asterisk: line 152: 23241 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal EXITSTATUS-128.
Automatically restarting Asterisk.
--
Everything is fine, before I install cdr_addon_mysql.so and modifying
cdr_mysql.conf. I use asterisk 1.6.2.1, asterisk-addon 1.6.2.0, Centos
5.3
Thanks in advance.
Quyps
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Re: [asterisk-users] How to log into separate file
On Wed, 2010-04-07 at 21:37 -0700, Steve Edwards wrote: On Thu, 8 Apr 2010, Pham Quy wrote: I want to have a separate file to log what i need for my dialplan without all output from Asterisk. By this way, i can easily to trace problems caused by my dialplan. You can control how much and where Asterisk logs by fiddling with logger.conf. Personally, I like logging through syslog so I can configure all of my Asterisk hosts to send all of their logging to a central log host. I also use syslog() in my AGIs so everything is in the same place. For production, I use: syslog.local0 = error When I'm debugging something, I'll crank up the volume like: syslog.local0 = debug,dtmf,error,event,info,\ notice,verbose,warning If this doesn't give you the level of logging you need, you could always call an AGI or do something really ugly like: exten = *,n,system(logger -t foo mumble) Thanks, I have another question. Does Asterisk support to define new logging level? Thanks in advance, Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to log into separate file
Hi all, I want to have a separate file to log what i need for my dialplan without all output from Asterisk. By this way, i can easily to trace problems caused by my dialplan. How can i do that? Thanks in advance. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk a-law header missing?
Hi all, I records a audio file as alaw format, i tried to play it with several audio player such as VLC, GNOME player, QUicktime,..but have no success. Do you know which player can play alaw file? I have a though that generated alaw file by asterisk seem not have header, therefore none of my tools can play it? Do you guys have any idea about this? Thanks in advance, Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asynchronous play music
Hi,
Thanks a lot Steve, Im gonna try it in Asterisk 1.6.
Quyps
On Tue, 2010-03-30 at 12:30 -0700, Steve Edwards wrote:
On Tue, 30 Mar 2010, Pham Quy wrote:
Is there anyway to catch DTMF keypress while a music file is playing
without stop the music?
On Mon, 29 Mar 2010, Steve Edwards wrote:
Have you tried externivr(). I've never used it, but it looks
interesting.
I hate it when people post about things they know nothing about...
So I wrote My First ExternalIVR (mfe.c). I'm a 1.2 Luddite, so things may
be a bit different for the Asterisk version you are using. Comparing the
documentation, it looks like a lot of useful commands have been added.
The first question I had was Where do you put the executable? The
documentation is silent on this point. ASTAGIDIR seemed a likely guess.
Guess not. It looks like it has to be an absolute path. Inconvenient, but
it works.
The actual protocol is trivial, way simpler and limited than AGI.
Here's a snippet that shows how to play a file and then handle the key
press events.
// play a file
printf(S,demo-congrats\n);
fflush(stdout);
// read events
while (NULL != fgets(event, sizeof(event), stdin))
{
syslog(LOG_ERR, event);
if ('#' == *event)
{
break;
}
}
(The full source is at http://www.sedwards.com/mfe.c)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Asynchronous play music
Hi all, Is there anyway to catch DTMF keypress while a music file is playing without stop the music? Thanks, Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting while playback
I think you would be more successful and have more control if you wrote it as an AGI. Then you could set a timer that would interrupt the process and you could do what you like from there (hangup?). I think you are asking too much of the dialplan. I would tend to leap into an AGI also, but did you try setting an absolute timeout? Externivr() may also be a good approach. Hi, I've been searching for some example of ExternIvr(), but didnt find any. Can you point where can i find example of how to use ExternIvr() Thanks in advances Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
Hi all, What i want to do is something like karaoke. when caller call to asterisk, a music song is played while caller sings. His or her voice will be recorded and mixed with music. To do that i use MixMonitor(). But i enable user to select a part of song to be recorded (monitored) as following: he press '*' to start recording. There are two way to stop recording: he press '#'to stop recording or it will be stop automatically after 60 seconds. My question is: how can i implement counting down 60s? ps: sorry for my english, thank you all for any help Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Time counting while playback
Hi all,
This question has been asked for days, I think that would be more
comprehensible if i post it in a new thread.
What i want to do is something like karaoke. when users call to
asterisk, a music song is played while caller sings. Their voice
will be recorded and mixed with the music. To do that i used
MixMonitor() and Playback() applications.
I also want to enable users to select a part of song to be recorded
(monitored) for example: Users press '*' to start recording. For
stopping record, there are two ways: (1) he press '#'to stop recording
OR it will be stopped (stop MixMonitor) AUTOMATICALLY after 60 seconds.
How can I count down 60s? MixMonitor app doesnt have any time out
argument.
I detect '#' using Read() app as following
[ivr-test]
exten = test,1,Answer()
exten = test,n,Wait(2)
exten = test,n(prompt),Read(digit,hello-world,1,,3,2)
exten = test,n,NoOp(Input digit - $[${digit}])
exten = test,n,GotoIf($[${digit} = 1]?one,1)
exten = test,n,GotoIf($[${digit} = #]?sharp,1)
exten = test,n,GotoIf($[${digit} = ]?nokey,1)
exten = test,n,Goto(prompt)
exten = test,n,Hangup()
exten = one,1,NoOp(1 pressed)
exten = one,n,Hangup()
exten = sharp,1,NoOp(You press # )
exten = sharp,n,HangUp()
exten = nokey,1,NoOp(No key pressed)
exten = nokey,n,Hangup()
---
But it couldnt read #, key '#' have recognized as NoKey
ps: sorry for my english
Quyps
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Re: [asterisk-users] Time counting down and # detect
Hi I made a conclusion too soon, i couldnt count down while music was playing back. I'm running out of time, really need help. Thanks QuyPs On Sat, 2010-03-13 at 08:42 +0700, Pham Quy wrote Here again, the script should be described as - Caller call to a number - Asterisk answer, play back music and start MONITORING as following + User press * to start MONITORING + Record is finished if: + User press # + OR message duration reach 60 second + Hangup Quyps On Sat, 2010-03-13 at 08:36 +0700, Pham Quy wrote: Hi Gordon, What i'm doing now is that something like karaoke. While music is playing back, caller voice is being record by the way i mentioned earlier. I should give you the whole picture of what i'm doing. I did google for it, and Monitor() function seem to be the best choice to do that. I would prefer using Record() if somehow i could play back music while recording. Thanks, Quyps On Fri, 2010-03-12 at 10:43 +, Gordon Henderson wrote: On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Time counting down and # detect
Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
Hi Gordon, What i'm doing now is that something like karaoke. While music is playing back, caller voice is being record by the way i mentioned earlier. I should give you the whole picture of what i'm doing. I did google for it, and Monitor() function seem to be the best choice to do that. I would prefer using Record() if somehow i could play back music while recording. Thanks, Quyps On Fri, 2010-03-12 at 10:43 +, Gordon Henderson wrote: On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting down and # detect
Here again, the script should be described as - Caller call to a number - Asterisk answer, play back music and start MONITORING as following + User press * to start MONITORING + Record is finished if: + User press # + OR message duration reach 60 second + Hangup Quyps On Sat, 2010-03-13 at 08:36 +0700, Pham Quy wrote: Hi Gordon, What i'm doing now is that something like karaoke. While music is playing back, caller voice is being record by the way i mentioned earlier. I should give you the whole picture of what i'm doing. I did google for it, and Monitor() function seem to be the best choice to do that. I would prefer using Record() if somehow i could play back music while recording. Thanks, Quyps On Fri, 2010-03-12 at 10:43 +, Gordon Henderson wrote: On Fri, 12 Mar 2010, Pham Quy wrote: I figured that out, i can use monitor() function to record and using a loop to count down 60s. But I dont think it is best solution, any suggestion is appreciated. And still, how can i capture '#'? Have you reied reading the manual, or the wiki, or even just googling for asterisk recording? You'll find the Record() application will do what you need to do regarding time and #. Gordon On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote: Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to create a dummy call
On Thu, 2010-03-04 at 01:48 -0600, Tilghman Lesher wrote: On Wednesday 03 March 2010 22:20:40 Pham Quy wrote: It maybe not clear that what i'm going to do. What i want to do is that enable user to call to a number then a background music will be played and he/she sing to mobilephone, the voice will be recorded and synchronized with the music. Any idea? There is an approach which using Monitor and Meetme application, it however need to throw an extra call to playing music, and this call should be thrown automatically by Asterisk. You really don't need to generate any call at all. Just Answer, Monitor, and Playback the sound file. Monitor will take care of mixing the sound file and the user's voice. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org It works perfectly, thanks you all -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play an audio file from a remote host
hi all, We going to implement a music service which enable user to playback a song by dialing to a service number. The problem is that the amount of data is huge so we have to plae it on an different server which is connected to the asterisk's via internet. Does asterisk support playing a audio file from an resource locate in a remote host? Please help, Thanks, Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to create a dummy call
Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to create a dummy call
Hi all, It maybe not clear that what i'm going to do. What i want to do is that enable user to call to a number then a background music will be played and he/she sing to mobilephone, the voice will be recorded and synchronized with the music. Any idea? There is an approach which using Monitor and Meetme application, it however need to throw an extra call to playing music, and this call should be thrown automatically by Asterisk. Again, any idea? Please help, thanks Quyps On Thu, 2010-03-04 at 10:37 +0700, Pham Quy wrote: Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-users] how to create a dummy call
Hi all, What i'm going to do is that enable caller sing while playing a background music. My approach is using Monitor and Meetme app. Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music. Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to play background music during record
Hi all, The question has already asked here, http://www.mail-archive.com/asterisk-users@lists.digium.com/msg98176.html but it's been two years since then, so is there any better solution with latest release version? Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ivvr with asterisk
Thanks all, Before purchasing any device i want to make some prototype of IVVR, is it possible to use asterisk to build an IVVR with softphones (such as SIP softphone)? and Is there any example about these? Quyps On Sat, 2010-01-23 at 11:44 +0530, mtha...@gmail.com wrote: Quyps, It looks like you mis-read the picture. Asterisk is the core, it need to be there regardless you use FreePBX or Tribox. FreePBX is a GUI web interface to manage asterisk. Itself is not an IP-PBX. Trixobx, still based on the Asterisk + freePBX, adds some more additional applications based on the community feed back and requirement. Trixbox is an easy go, but there may be some unwanted stuff with it. elastix.org is also a nice package, give it a try. Regards MT Kondela kevesystems.com On Sat, Jan 23, 2010 at 7:32 AM, Pham Quy qu...@vega.com.vn wrote: Hi all, First I'm very new. I want to build an Interactive Video-voice Response system. There is number of choice I have found so far: FreePBX, TriBox, Asterisk. Which is the best in my case? and what do i need to build such IVVR system? Thanks. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ivvr with asterisk
Hi all, First I'm very new. I want to build an Interactive Video-voice Response system. There is number of choice I have found so far: FreePBX, TriBox, Asterisk. Which is the best in my case? and what do i need to build such IVVR system? Thanks. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
