similar to: Any way of "flattening out" 2 channels back into one?

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]

Please point me to samples of popping and wiping hangup handlers. I don't need to use the values returned; I just need to clear any handlers before I push a new one. It's not clear at https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how to provide vars on the right-hand side. Cheers, David -------------- next part -------------- An HTML attachment was scrubbed...

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

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

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, 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:

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

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

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 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

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 --------------

This has been super-helpful, Eric. However, the handleHangupByPeer priorities below are still not run when the peer hangs-up. The last line in the cli when the peer hangs-up is still: Strict RTP learning complete - Locking on source address (Although sometimes there is also: Retransmission timeout reached on transmission) same =>

Thanks, Eric. I just tried a hangup handler, but it's showing a similar problem: When the peer hangs-up, the hangup handler is not invoked and the caller channel remains open. same => n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount} + 1]) same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCallerOrPeer,doesntMatter,1(args)) same =>

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Hi, I have the following code that operates when a channel is hung-up: [record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return() Before the dial a hangup handler is registered: Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1) The routine is called and the variables are being set, however not on the channel's CDR which made the call. I believe this

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

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne