search for: hangup

...key #, the phone which is connected to BB was constantly trying to connect to AA. Here is the Dialplan configuration: Asterisk server number one (AA) All outgoing calls need a 5555 prefix in front of the number [buster] ; btc trunk exten => _9.,1,DIAL(SIP/btctrunk/${EXTEN}) exten => _9.,n,Hangup() ; soft ; caller-id SOFT ;exten => _X.,1,Set(CALLERID(num)=015555${CALLERID(num)}) exten => _X.,1,Set(CALLERID(num)=0158631${CALLERID(num)}) exten => _X.,n,DIAL(SIP/softnet/555501${EXTEN}) exten => _X.,n,Hangup() ; popravek 0 ;exten => _0.,1,Set(CALLERID(num)=018109${CALLERID(num)...

hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten => 20000,1,Dial(SIP/20000,30,Ttm) exten => 20000,2,Hangup exten => 20000,102,Voicemail(20000) exten => 20000,103,Hangup exten => 20100,1,Dial(SIP/20100,30,Ttm) exten => 20100,2,Hangup exten => 20100,102,Voicemail(20100) exten => 20100,103,Hangup exten => 20200,1,Dial(SIP/20200,30,Ttm) exten => 20200,2,Hangup exten => 202000,10...

Hello asteriskers, I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> prompt, and found references on using the command "soft hangup <SIP/channel>", but as you can see below, the "soft hangup" command does not seem to exist, and there is no mention about it in the UPGRADE*.txt documents. Can an...

Hello, you know , when a call setup, either caller hangup first or callee hangup first , the hangupcause will set to 16(means Call Clearing Causes) My question is how could i identify whether the caller or callee hangup the phone first? Best Regards! -- Thanks for your supporting, have a nice day. Sucan

hi, I hope someone can help me:-) we’ve got a freepbx server. there are 2 special extensions (2001, 2002). if someone calls this extensions (or a call is forwarded to these extensions) and these extension hangup (not the caller party), then we’d like to put the calls back into a queue (1000) and wouldn’t like to hangup. I read your description about hangup hooks: https://community.freepbx.org/t/hooking-for-fun-and-income/57718 but still not able to implement it:-( what I’ve done: * found out in a hard w...

...m trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm) exten => *0,3,Playback(fcopba1) exten => *0,4,Hangup() include => extentions include => pasvalide [extentions] exten => 1001,1,Dial(SIP/sipchan1001,10) exten => 1001,2,Voicemail(u1001) exten => 1001,3,Hangup() ex...

Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-00000007 Usage: channel request hangup <channel>|<all> Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If '...

...DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:46 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src=8, dst=908 2005-05-20 06:55:46 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src=8, dst=908 2005-05-20 06:56:27 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src=3, dst=558 2005-05-20 06:56:28 DEBUG[18936]: Stopping retransmission on '6efad2d21ca04d653c1c24a25ada1449@192.168....

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code returned to the calling party is...

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

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 => n(callPeer),Set(GLOBAL(...

...alleridonzaptransfer=yes ;callprogres=yes ;echocancelwhenbriged=yes rxgain=10.2 txgain=1.8 signalling=fxs_ks channel=>80 ;------------------------------ context=service ;------------------------------ group=7 callgroup=7 ;echocancelwhenbriged=yes callerid=asreceived answeronpolarityswitch=yes hanguponpolarityswitch=yes ;cidstart=ring ;cidsignalling=v23 sendcalleridafter=2 rxgain=5.5 txgain=7.2 signalling=fxs_ks channel=>81 ;------------------------------ context=service_out ;------------------------------ group=7 callgroup=7 ;echocancelwhenbriged=yes callerid=asreceived ;cidstart=ring ;c...

I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. I send SIP/.... to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,P...

I have a customer who named their endpoint to include a space (example, 1003 a) >From the CLI, I want to hangup a channel on this endpoint >From core show channels concise, I see the channel name includes the space PJSIP/1003 a-00000002 I realize the space is interpreted as an argument separator, so my first attempt below doesn't work. I have tried the following and all fail. hangup request PJSIP/1...

...been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My problem seems to be with the BT102s. I can't seem to get them to hangup when issued a Hangup(). Asterisk's logs show it's has hung up, but I get a busy signal on the phones themselves. This was also happening under firmware 1.0.5.16. Configuration listed below; all of them are called via the #include statement: <<<snip>>> my.sip.conf...

Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten => s,5,Dial(SIP/302,25) exten => s,6,Hangup exten => s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is noth...

I am using the new 1.8.5 and I just found out that Asterisk won't record the call if the call just hangup. I did a test like this: exten => 1009, 1, Hangup() Then I called 1009: == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [1009 at init-1005:1] Hangup("SIP/1007-0000003c", "") in new stack == Spawn extension (init-1005, 1009, 1) exited non-ze...

Hi list, n00b question, but I can't figure it out: [callthrough] exten => _+X.,1,NoOp(nothing here) #include "blockedall.conf" exten => _+X.,n(hangup),Hangup exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" = "anonymous"]?nocli:cli) ... more stuff that is handling the call ... I'm putting CLIs that I don't want to be able to call my system into blockedall.conf: exten => _+X.,n,GotoIf($["${CALLERID(...

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). > > > I want to have my custom hangup-handler be run only once and when >...

...e tried various X100P (original Digium) cards, various phone lines and just about every possible combination of parameters in zaptel.conf (from every feature off to most features on) but without any success. The same hardware was working fine with CVS from about a month ago. The problem of false hangups really needs to be fixed. A false hangup is NEVER EVER acceptable in an office environment. On the other hand, a call that doesn't hangup even if the remote party has already hung up is ALWAYS acceptable. Therefore, if the software is not capable of detecting hangups properly, then why not pro...