[asterisk-users] how to load our own .wav sound files in the dial plans for playback
Hi, i am trying to add my own sound file in the asterisk dial plan extension for playback option , i dont no where to put the file and how to give the path in extension file and all so is need that the sound file should be convert in asterisk as .wav file??? regards Upendra -- _ -- 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 load our own .wav sound files in the dial plans for playback
On Sat, 8 Sep 2012, upendra wrote:
i am trying to add my own sound file in the asterisk dial plan extension
for playback option , i dont no where to put the file and how to give
the path in extension file and all so is need that the sound file should
be convert in asterisk as .wav file???
A command similar to:
sox\
${INPUT}\
--bits 16\
--channels 1\
--encoding signed-integer\
--rate 8k\
${OUTPUT}
will convert the file to a format Asterisk will be happy with.
When you specify a file to be played, for example, using 'playback()', you
specify the file's path. If you don't specify an absolute path, the path
is relative to the directory set in asterisk.conf.
One last 'oops' I still do is forgetting that Asterisk likes to choose the
'file type' (even if you only have .wav) so don't specify it.
So, for example, if you had a file named 'my-first-wave-file.wav' in a
sub-directory of Asterisk's 'sounds' directory named 'upendra' you could
play the file with either:
playback(upendra/my-first-wave-file)
playback(/var/lib/asterisk/sounds/upendra/my-first-wave-file)
www.voip-info.org is an excellent (if somewhat dated) resource for
Asterisk questions. I'm sure asterisk.org and ATFOT.pdf are also
excellent resources. I started with voip-info.org, so it's my 'go to'
resource.
--
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] One leg in a conference and adjusting stream volume of other leg
Matthew, Johan, everyone,
I got it to work! :) (With the help of a guy I hired via freelancer.com)
Time to share something with the community, so here are the pieces you
need to create an open conference, where the users will be in the same
conference, and at the same time will listen to an individual MP3 stream
in the background, depending on which extension a user dials. Also, the
volume of the stream is adjustable for each user separately via DTMF
1+2, and the speech in the conference is also adjustable individually
via DTMF 4+5 and 7+8 (this is plain ConfBridge, no magic there).
Extensions 01 or 02 is what the user dials. If dialed 01, user will
listen to stream 1, if dialed 02, user will listen to stream 2, but both
users will be in the same conference and will not hear each others
music, but only each others speech.
extensions.conf:
[macro-mohvolumeup]
exten = _.,1,NoOp(Increasing MOH volume...)
exten = _.,n,NoOp(...for extension ${sipexten} )
exten = _.,n,System(/var/lib/asterisk/agi-bin/mohvolume.php ${sipexten} up)
[macro-mohvolumedown]
exten = _.,1,NoOp(Decreasing MOH volume...)
exten = _.,n,NoOp(...for extension ${sipexten} )
exten = _.,n,System(/var/lib/asterisk/agi-bin/mohvolume.php ${sipexten}
down)
[radio-chatfire]
exten = go-conference,1,Answer()
exten = go-conference,n,NoOp(MOH class is ${mohclass})
exten = go-conference,n,System(/var/lib/asterisk/agi-bin/playmoh.php
${sipexten} ${mohclass})
exten =
go-conference,n,ConfBridge(11*48*79*32,,chatfire-public,chatfire-public-menu)
; chat, stream 1
exten = 01,1,NoOp(Dial)
exten = 01,n,Set(__sipexten=${CHANNEL})
exten = 01,n,Set(__DYNAMIC_FEATURES=mohvolumeup#mohvolumedown)
exten = 01,n,Set(__mohclass=chatfire-1)
exten = 01,n,NoOp(MOH class is ${mohclass})
exten = 01,n,Dial(Local/go-conference@radio-chatfire,,)
; chat, stream 2
exten = 02,1,NoOp(Dial)
exten = 02,n,Set(__sipexten=${CHANNEL})
exten = 02,n,Set(__DYNAMIC_FEATURES=mohvolumeup#mohvolumedown)
exten = 02,n,Set(__mohclass=chatfire-2)
exten = 02,n,NoOp(MOH class is ${mohclass})
exten = 02,n,Dial(Local/go-conference@radio-chatfire,,)
exten = 5,1,Answer()
exten = 5,n,Set(VOLUME(TX,p)=-3)
exten = 5,n,NoOp(MOH class final is ${mohclass_final})
exten = 5,n,MusicOnHold(${mohclass_final})
[whisper-chatfire]
exten = do_chanspy,1,NoOp()
exten = do_chanspy,n,Set(DB(moh_${sipexten}/channel)=${CHANNEL})
exten = do_chanspy,n,ChanSpy(${sipexten},${chanspyoption})
exten = do_chanspy,n,Hangup()
exten = do_moh,1,NoOp(Dial)
exten = do_moh,n,Set(__mohclass_final=${mohclass_play})
exten = do_moh,n,Dial(Local/5@radio-chatfire)
manager.conf:
[general]
enabled=yes
port=5038
bindaddr=127.0.0.1
[manager]
secret=kk
allow=0.0.0.0/0.0.0.0
read =
all,system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,originate
write = all,system,call,agent,user,config,command,reporting,originate
features.conf:
[applicationmap]
mohvolumeup = 1,self/caller,Macro,mohvolumeup
mohvolumedown = 2,self/caller,Macro,mohvolumedown
confbridge.conf:
[chatfire-public-menu]
type=menu
4=increase_listening_volume
5=decrease_listening_volume
7=increase_talking_volume
8=decrease_talking_volume
[chatfire-public]
type=user
announce_user_count=yes
dsp_drop_silence=yes
musiconhold.conf:
[chatfire-1]
mode=custom
application=/var/lib/asterisk/mohstream-chatfire-1.sh
[chatfire-2]
mode=custom
application=/var/lib/asterisk/mohstream-chatfire-2.sh
/var/lib/asterisk/agi-bin/playmoh.php:
#!/usr/bin/php -q
?php
$sipexten = $argv[1];
$mohclass_play = $argv[2];
$wrets = ;
$amiusername = 'manager';
$amisecret = 'kk';
ob_implicit_flush(true);
$chanspyoption = qWEws;
// Some of these options dont seem to exist, but whatever, it works :)
ob_implicit_flush(true);
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 0);
$wrets = fread($socket,30);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: $amiusername\r\n);
fputs($socket, Events: off\r\n);
fputs($socket, Secret: $amisecret\r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, Channel: Local/do_chanspy@whisper-chatfire\r\n);
fputs($socket, Exten: do_moh\r\n);
fputs($socket, Context: whisper-chatfire\r\n);
fputs($socket, Priority: 1\r\n);
fputs($socket, Variable: chanspyoption=$chanspyoption\r\n);
fputs($socket, Variable: sipexten=$sipexten\r\n);
fputs($socket, Variable: mohclass_play=$mohclass_play\r\n\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
while (!feof($socket)) {
$wrets .= fread($socket,8192 );
}
fclose($socket);
?
/var/lib/asterisk/agi-bin/mohvolume.php:
#!/usr/bin/php -q
?php
$sipexten = $argv[1];
$command = $argv[2];
if ($command == up) {
$dtmftone = *;
} elseif ($command == down) {
$dtmftone=#;
}
$wrets = ;
$amiusername = 'manager';
$amisecret = 'kk';
ob_implicit_flush(true);
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 0);
$wrets = fread($socket,30);
//AMI Login
fputs($socket, Action: Login\r\n);
Re: [asterisk-users] Calls from talkonaut to pstn Phone
...@siena.edu, McHugh, Kim kmch...@siena.edu Message-ID: de01b7d81028c145884d624c2d361f710137ddee3...@mb-2.siena.edu Content-Type: text/plain; charset=us-ascii We have a Rolm 9751 connecter to our asterisk box via a straight T1. The Rolm cannot do PRI. Has anyone figured out how to configure this link (probably on the Rolm side) to pass caller ID? Any Help or suggestions, aside from forklifting the Rolm, would be appreciated. Thanks, Edward Kohler Network Technician 101Hines Hall Siena College 515 Loudon Rd. Loudonville, NY 12211 518-783-2391 Fax 518-783-2590 ekoh...@siena.edumailto:ekoh...@siena.edu Siena College is a learning community advancing the ideals of a liberal arts education, rooted in its identity as a Franciscan and Catholic institution. CONFIDENTIALITY NOTICE: This e-mail, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure, or distribution is prohibited. If you received this e-mail and are not the intended recipient, please inform the sender by e-mail reply and destroy all copies of the original message. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20120907/a318f4fa/attachment-0001.htm -- Message: 3 Date: Sat, 8 Sep 2012 14:19:51 +0530 From: upendra uppi...@gmail.com Subject: [asterisk-users] how to load our own .wav sound files in the dialplans for playback To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: CAFogui8j2Ny3wChAu-t29PyPT=CnZqpedtAJzDn3R= lrc+a...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, i am trying to add my own sound file in the asterisk dial plan extension for playback option , i dont no where to put the file and how to give the path in extension file and all so is need that the sound file should be convert in asterisk as .wav file??? regards Upendra -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20120908/7c81ae11/attachment-0001.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 98, Issue 10 ** -- _ -- 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
