similar to: asterisk 1.2 and Dial with LIMIT_WARNING_FILE

Dear All, I tried to use 'L' option on my dial command. I set the x to 65000(65 seconds), y to 60000(60 seconds), z to 30000(30 seconds). The max calltime should be 65 seconds, and it will play "beep.gsm" at 60 seconds left. And repeat the beep at 30 seconds left. But the call will be hangup by system at 60 seconds left. In another word, when it plays warning file, the call

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

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

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]

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:

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

Thank you - that makes sense. I've seen something about swapping and optimizing channels on the console, but I didn't realise "optimize" meant "not do what you wanted". OK, so here's why I'm dialling anything at all: The first dial is because I MUST limit the incoming call to less than 60 minutes. The second dial, which carries the gH option, is because I

Hi, I?m playing with the CVS head time limiting at Dial application, it just works fine but the only problem is that the caller isn?t hearing the warning message. I?m using a Cisco 7960 as the caller and a Polycom 500 as the callee. The audio is passing through Asterisk: -- Executing Dial("SIP/8992-9712", "SIP/8988|20|L(10000:2000)") in new stack -- Limit Data: --

hi, i'm added another patch to chan_ss7 it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/ New in version 0.10.1 (community version) - support for more than 256 channels - zap style addressing http://download.seiros.ru/SeirosPBX/chan_ss7/ http://www.freevoice.cz/chan_ss7/chan_ss7-0.10.1.tar.gz md5sum a3ca3031f8f4ef96d505be3b297b47cc

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

Hello everyone, I am trying to apply this<http://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diff>patch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors: Hunk #1 FAILED at 704. Hunk #2 FAILED at 715. I have done the patch modifications manually in l4isup.c There is just one question, how do I pass the RB file-to-play on an SS7 channel via

Muchas gracias Miguel. Os pongo mi solución por si alguien necesita llamar a distintos tipos de sistema. setwd("C:/Users/usuario/Desktop/Pruebas/") x <- sessionInfo() sistema <- substr(x$R.version$system,1, 3 ) origen <- "Carp" destino <- "Carp235" switch(sistema, x86 = system(paste(Sys.getenv("COMSPEC"),"/c

I still see: @pkgdatadir@, etc. in the paths of the icecast.xml. Is this the used configuration file for the actual icecast ? Then the should be replaced by the actual directory names. Isn't it ? Henk -----Oorspronkelijk bericht----- Van: Icecast [mailto:icecast-bounces at xiph.org] Namens icecast-request at xiph.org Verzonden: vrijdag 31 augustus 2018 14:00 Aan: icecast at xiph.org

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Hello on Behalf Of idont know, Sangoma has a Media Gateway solution via SS7. They I believe are the only ones capable of connecting Asterisk via SS7. You may want to check them out. Heidi -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of idont know Sent: April 6, 2006 10:29 AM To: asterisk-biz@lists.digium.com

Hi, Has anyone tried using these flags for the Dial command? I set it to "no" but both parties can still hear the (beep) warning sound. - Wei

I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 =