search for: limit_warning_file

...tem at 60 seconds left. In another word, when it plays warning file, the call will be hangup. The answeredtime is only 5 seconds. Can anybody give me an idea for it? *********** extensions.conf *********************** [default] exten=> _+[1-9].,1,SetCallerID(22222222) exten=> _+[1-9].,2,Set(LIMIT_WARNING_FILE=beep) exten=> _+[1-9].,3,Set(LIMIT_TIMEOUT_FILE=beep) exten=> _+[1-9].,4,Dial(zap/g1/002${EXTEN:1}|60|L(65000:60000:30000)) exten=> _+[1-9].,105,Hangup ************ Log from CLI ******************************************************* -- Seeding '24012100' at 61.217.XXX.XXX:86...

Hi fellows.. I have 2 asterisk servers in which the following line exten => _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES) exten => _09049.,112,SetVar(LIMIT_WARNING_FILE=beep) exten => _09049.,113,Dial(${TYPE}${DESTINO}|30|L(30000:10000)) works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the end of the call. doesn't work on my Asterisk 1.2.13, it hungs 10 seconds before the end of the call, just when it has to beep both of them have t...

Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached) same => n,Dial(Local/s at root/n,3,L(3540000:60000)) same => n,Hangup() [root] exten => s,1,Verbose(1,Call to: ${CALLERID(name)} from: ${CALLERID(num)}) same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1) etc etc Works...

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk 14.1 Here's a bit of test dialplan, which works as expected and simulates exactly what I'm doing at the top of my large dialplan... [dial-pre-test] exten => s,1,NoOp() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) same => n,Dial(Local/s at dial-test,3,L(3540000:60000)) same => n,Hangup() [dial-test] exten => s,1,NoOp() same => n,Dial(Local/s at dial-dest,,gH) same => n,Playback(goodbye) same => n,Hangup() [dial-dest] exten => s,1,Answer() sam...

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

...sounds to the caller. + *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. + *LIMIT_TIMEOUT_FILE* - File to play when time is up. + *LIMIT_CONNECT_FILE* - File to play when call begins. + *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then the default behaviour is to announce ("You have [XX minutes] YY seconds"). o *f*: forces callerid to be set as the extension o...

...ee=no -- warning_freq=0 -- start_sound=UNDEF -- warning_sound=timeleft -- end_sound=UNDEF -- Called 8988 -- SIP/8988-6922 is ringing -- SIP/8988-6922 answered SIP/8992-9712 == Spawn extension (local, 8988, 1) exited non-zero on 'SIP/8992-9712' If I change the LIMIT_WARNING_FILE to something like 'beep' to use the usual beep.gsm file same results :( Any suggestions ? -- Juanjo sin .sig

Greetings, I have tried numerous ways to set the LIMIT_PLAYAUDIO_CALLER and LIMIT_PLAYAUDIO_CALLEE variables with no success. The default parameters never change. Has anyone had success changing the defaults? If so, how did you do it? Thanks, vcomp -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.....do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can?t be upgraded (policy

Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well as a lack of spy/whispering commands available via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

...; Here's a bit of test dialplan, which works as expected and simulates >> exactly what I'm doing at the top of my large dialplan... >> >> [dial-pre-test] >> exten => s,1,NoOp() >> same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) >> same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) >> same => n,Dial(Local/s at dial-test,3,L(3540000:60000)) >> same => n,Hangup() >> >> [dial-test] >> exten => s,1,NoOp() >> same => n,Dial(Local/s at dial-dest,,gH) >> same => n,Playback(goodbye) >>...

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------