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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.36s
  • Memory usage: 2.21MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 3.33

Asterisk cmd BackGround

Synopsis

Play a sound file while awaiting extension

Description

 Background(filename1[&filename2...][|options[|langoverride][|context]])

This application will play the given list of files while waiting for an extension to be dialed by the calling channel. To continue waiting for digits after this application has finished playing files, the WaitExten application should be used. The 'langoverride' option explicity specifies which language to attempt to use for the requested sound files. If a 'context' is specified, this is the dialplan context that this application will use when exiting to a dialed extension. If one of the requested sound files does not exist, call processing will be terminated.

Options

  • s - causes the playback of the message to be skipped if the channel is not in the 'up' state (i.e. it hasn't been answered yet.) If this happens, the application will return immediately.
  • n - don't answer the channel before playing the files
  • m - only break if a digit hit matches a one digit extension in the destination context

Returns

See http://bugs.digium.com/view.php?id=7835 for details of a crude patch to implement the return of ${BACKGROUNDSTATUS} ("FAILED" or "SUCCESS" in the same way Playback() does).

Wait for DTMF Input

If you want Asterisk to just wait for input without playing a sound file, see the WaitExten application.

Example

 ; Sample 'main menu' context with submenu
 exten => s,1,Answer
 exten => s,2,Background(thanks) ; "Thanks for calling. Press 1 for sales, 2 for support, ..."
 exten => 1,1,Goto(submenu,s,1)
 exten => 2,1,Hangup

See also



Asterisk | Asterisk Configuration | The Dialplan: extensions.conf | Dialplan Commands

Created by oej, Last modification by Paul Gillman on Fri 13 of Apr, 2007 [13:10 UTC]

Comments Filter

m option in 1.4

by jack on Thursday 25 of January, 2007 [19:09:04 UTC]
In my dialplan, I have the following:

exten => s,1,Background(${RECORDING}|m)
exten => s,n,Voicemail(${DID_NO})
exten => 0,1,Voicemail(${DID_NO})
exten => a,1,VoiceMailMain(${DID_NO})
exten => h,1,Hangup

In version 1.2, when I hit "0" during the playback, I will be directed to voicemail. But in verison 1.4, the call hangs up.

Jan 24 16:05:37 DTMF5754: channel.c:2148 __ast_read: DTMF begin '0' received on SIP/5060-08c53e68
Jan 24 16:05:37 DTMF5754: channel.c:2128 __ast_read: DTMF end '0' received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on 'SIP/5060-08c53e68'
-- Executing h@play_recording:1 Hangup("SIP/5060-08c53e68", "") in new stack
== Spawn extension (play_recording, h, 1) exited non-zero on 'SIP/5060-08c53e68'


Does anyone tell me why this is happening?


Sound formats

by Alex on Wednesday 05 of April, 2006 [02:45:16 UTC]
Can other file formats be used instead of gsm for audio files? What about wav or mp3? If not, what's the recommended way of converting an existing audio file to the gsm format? Thanks!

Edit:
Nevermind. Just had a look at http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

String Multiple Files Together

by abombss on Tuesday 22 of March, 2005 [06:29:35 UTC]
On the CVS version you can play multiple messages back to back by using an & to seperate the file names.

Ex: prompt,1,Background(press-1&for-sales&press-2&for-billing)

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