Re: [asterisk-users] how to play music when dial fail or time out
Thanks Matt, can I transform the format in dial plan? On Wed, May 11, 2011 at 11:24 AM, Matt Riddell li...@venturevoip.comwrote: On 11/05/11 3:11 PM, John Wu wrote: Hi Enrico thanks I do what u said but meet this problem: [May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File fail.wav does not exist in any format [May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to open fail.wav (format 0x2 (gsm)): No such file or directory [May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/IMSI460020656177633- for fail.wav When you playback a file in Asterisk you don't provide the extension. So you'd do Playback(fail) rather than Playback(fail.wav) That way if you have 10 files all called fail (i.e. fail.wav, fail.gsm, fail.ulaw, fail.alaw etc etc) it will pick the one which matches the phone calls. For example in the above example you were making a call in the GSM format but it couldn't find a file called fail.wav.gsm so it looked for fail.wav.* and couldn't find anything. Basically just drop the extension. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 play music when dial fail or time out
Thanks Matt the problem is solved. On Wed, May 11, 2011 at 11:24 AM, Matt Riddell li...@venturevoip.comwrote: On 11/05/11 3:11 PM, John Wu wrote: Hi Enrico thanks I do what u said but meet this problem: [May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File fail.wav does not exist in any format [May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to open fail.wav (format 0x2 (gsm)): No such file or directory [May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/IMSI460020656177633- for fail.wav When you playback a file in Asterisk you don't provide the extension. So you'd do Playback(fail) rather than Playback(fail.wav) That way if you have 10 files all called fail (i.e. fail.wav, fail.gsm, fail.ulaw, fail.alaw etc etc) it will pick the one which matches the phone calls. For example in the above example you were making a call in the GSM format but it couldn't find a file called fail.wav.gsm so it looked for fail.wav.* and couldn't find anything. Basically just drop the extension. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 play music when dial fail or time out
Hi Enrico thanks I do what u said but meet this problem: [May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File fail.wav does not exist in any format [May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to open fail.wav (format 0x2 (gsm)): No such file or directory [May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/IMSI460020656177633- for fail.wav fail.wav is the sound which I want to playback, from the message this file can not be found, I put it under /var/lib/asterisk/sounds/ On Mon, May 9, 2011 at 9:37 PM, Enrico Cicconi enrico.cicc...@rdmnet.itwrote: Hi, have you tried to manage all with dialplane ? just an example: [incoming] **exten = s,1,Dial (SIP/your_called_party,20) exten = s,n, Playback(music_message) . In the first step the call is redirect to the configured called party and if without answer (busy, not logged, not answered) ... ... in the second step a music is played. You can also do other kind of job instead of 'playback' if you need. Hoping to have helped you, have a nice day Enrico www.rdmnet.it http://www.rdmnet.it/asterisk/104-asterisk-in-pillole-impostare-il-dialplan.html Il 09/05/2011 09:41, John Wu ha scritto: Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -- _ -- 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 play music when dial fail or time out
thanks Dovid.
On Mon, May 9, 2011 at 5:30 PM, Dovid Bender asteriskus...@dovid.netwrote:
John,
You want to do it only after it fails ?
If so you can do something like.
Exten = _X., 1, Dial(SIP/${EXTEN}@PEER SIP/$%7BEXTEN%7D@%3CPEER%3E,20
)
Exten = _X., 2, GotoIf($[${DIALSTATUS} = ANSWER]?10)
Exten = _X., 4, MusicOnHold()
Exten = _X., 10, Hangup
- Original Message -
*From:* John Wu jwjo...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Monday, May 09, 2011 10:41
*Subject:* [asterisk-users] how to play music when dial fail or time out
Hi all,
I need to support this feature. When caller dial if the dial fail or no
answer from the
called number then play a music. So how to achieve that?
Thanks!
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[asterisk-users] how to play music when dial fail or time out
Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -- _ -- 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 let the call play audio when the dial fail
Hi all, I want to play an audio hint to caller when his dial fail rather than the current sound dodo. The caller use asterisk to do the call so I want to setup this asterisk to achieve play specific audio when the dial fail or time out. How to setup to achieve that? Thanks! -- _ -- 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] why asterisk sip dial procedure take long time
Hi all I use asterisk connect sip. I dial a phone number which is outside network so asterisk will dial the phone through sip server. But this procedure take too long between dial and ringing (about 40s). Any way to reduce the time it take? Regards! -- _ -- 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] can anyone tell me how to set asterisk to record all phonecall
Hi Jeremy
Thanks for your reply, I did what u said, but still can not work. the
/var/spool/asterisk/monitor
do not have any file. Attachment is my extensions.conf
On Sat, Mar 5, 2011 at 12:37 PM, Jeremy Kister asterisk...@jeremykister.com
wrote:
On 3/4/2011 9:49 PM, John Wu wrote:
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
this is one way it can be done
make sure you have 'lame' installed.
- in your extensions.conf:
[global]
VSA=/var/spool/asterisk
[outbound-or-wherever-you-dial]
exten = _XXX,1,Macro(Snoop,${EXTEN})
exten = _XXX,n,Dial(SIP/${EXTEN},${TIMEOUT})
exten = _XXX,n,StopMixMonitor
; above in case you're in some loop Dial fails,
; e.g., swift+monitor crash asterisk
[macro-Snoop]
; ${ARG1} channel
exten = s,1,GotoIf($[${SNOOPING} = 1]?snooping)
exten = s,n,Set(SNOOPING=1)
exten = s,n,Set(=${STRFTIME(${EPOCH},,%Y)})
exten = s,n,Set(MM=${STRFTIME(${EPOCH},,%m)})
exten = s,n,Set(DD=${STRFTIME(${EPOCH},,%d)})
exten = s,n,Set(HMS=${STRFTIME(${EPOCH},,%H%M%S)})
exten = s,n,Set(FILENAME=${HMS}-${CALLERID(num)}-${ARG1}-${UNIQUEID})
exten = s,n,Set(MIXMON_ARGS=mkdir -p ${VSA}/monitor/${}/${MM}/${DD}
nice -n 19 /usr/local/bin/lame --silent --resample 11.025 -b 16 -t -m m
${VSA}/monitor/${FILENAME}.wav
${VSA}/monitor/${}/${MM}/${DD}/${FILENAME}.mp3 rm -f
${VSA}/monitor/${FILENAME}.wav)
exten = s,n,MixMonitor(${FILENAME}.wav,,${MIXMON_ARGS})
exten = s,n(snooping),NoOp(snooping on ${CHANNEL})
that'll end up putting a mp3 of the call in
/var/spool/asterisk/monitor//MM/DD/HHMMSS-CALLERID.mp3
don't forget any legal issues you might have to work around, recording the
fact that you declared the message is being recorded.
--
Jeremy Kister
http://jeremy.kister.net./
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extensions.conf
Description: Binary data
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Re: [asterisk-users] can anyone tell me how to set asterisk to record all phonecall
Thanks Steve, But I do not have many time to read asterisk usage. I will read it when I got free time. I find other ones use [macro-record-enable] to achieve auto record phone call. Can I directly use it? and how to debug the configure file? On Mon, Mar 7, 2011 at 10:16 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 7 Mar 2011, John Wu wrote: Thanks for your reply, I did what u said, but still can not work. the /var/spool/asterisk/monitor do not have any file. Attachment is my extensions.conf I don't think Jeremy intended for you to copy his example literally. Do you really have your endpoints pointed at '[outbound-or-wherever-you-dial]?' I suggest you take a step back and read 'Asterisk: The Future of Telephony' to get a bit more insight into how Asterisk works. You can buy a paper copy or google for the free PDF. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] can anyone tell me how to set asterisk to record all phonecall
In my asterisk config files, extensions_additional.conf do not exist so should I set [macro-record-enable] in extensions_additional.conf or in extensions.conf? My version is 1.8.0beta5 On Mon, Mar 7, 2011 at 10:45 AM, John Wu jwjo...@gmail.com wrote: Thanks Steve, But I do not have many time to read asterisk usage. I will read it when I got free time. I find other ones use [macro-record-enable] to achieve auto record phone call. Can I directly use it? and how to debug the configure file? On Mon, Mar 7, 2011 at 10:16 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 7 Mar 2011, John Wu wrote: Thanks for your reply, I did what u said, but still can not work. the /var/spool/asterisk/monitor do not have any file. Attachment is my extensions.conf I don't think Jeremy intended for you to copy his example literally. Do you really have your endpoints pointed at '[outbound-or-wherever-you-dial]?' I suggest you take a step back and read 'Asterisk: The Future of Telephony' to get a bit more insight into how Asterisk works. You can buy a paper copy or google for the free PDF. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] can anyone tell me how to set asterisk to record all phonecall
Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards! -- _ -- 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
