search for: hangsup

Displaying 20 results from an estimated 25 matches for "hangsup".

2007 May 09
3
The 'h' extension problem
Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling from out of our asterisk system). I understand if U1 hangsup then there is no channel to execute h extension, but is it possible to ex...
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
We're using Asterisk 14.7.6 and I have a dialplan that ends like this: same => n,Dial(SIP/${EXTEN:0:4}@peer1) same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) same => n,Hangup() When peer1 hangsup, the priorities after the Dial are executed fine. But when the caller hangsup during the Dial, the cleanup steps aren't done. Why? I did read "Note that on a successful connection, in the absence of the g and G modifiers (below), the Dial command does not return to allow execution of furt...
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
...33:26, David P wrote: > > > We're using Asterisk 14.7.6 and I have a dialplan that ends like this: > > > > same => n,Dial(SIP/${EXTEN:0:4}@peer1) > > same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) > > same => n,Hangup() > > > > When peer1 hangsup, the priorities after the Dial are executed fine. But > > when the caller hangsup during the Dial, the cleanup steps aren't done. > > Why? > > > > I did read "Note that on a successful connection, in the absence of the g > > and G modifiers (below), the Dial c...
2006 Apr 07
1
transfer call after advise
...\r\n Priority: 1\r\n\r\n this works fine (maybe the sintax now isn't correct... but it works), but my problem is that the call is immediately transferred to 500. I'd like if: 1 - 200 calls 400 2 - 400 want to transfer the call to 500 3 - 400 asks 500 if 500 wants to talk with 200 if 500 hangsup 200 still talk with 400 if 400 hangsup 200 talks now with 500 is it possible? thanks nik
2012 Oct 09
2
Asterisk sends wrong fxs 'Idle' hints
Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the "Idle" state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched of...
2005 Mar 13
2
sending a DTMF tone before hangup
...en you hang up the grandstream phone, thus telling the rocom unit to end the call ?? I did create a custom Playtones item to generate the 3 DTMF tone and added lines to my extensions.conf file as follows : [door] exten => s,1,Dial (SIP31,15) exten => s,2,Playtones(dtmf) However the call hangsup before trying to play the DTMF tone. Any ideas ?? I do here of people complaining that there are too many easy questions on the list so perhaps this will tax your brains :-) Thanks Nigel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/p...
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
...'re using Asterisk 14.7.6 and I have a dialplan that ends like >> this: >> > > same => n,Dial(SIP/${EXTEN:0:4}@peer1) >> > same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) >> > same => n,Hangup() >> > > When peer1 hangsup, the priorities after the Dial are >> executed fine. But >> > when the caller hangsup during the Dial, the cleanup steps aren't >> done. >> > Why? >> > > I did read "Note that on a successful connection, in the >> absence of t...
2005 Jun 22
3
Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
Hi, I'm pulling my hair down and getting bold :-) ..... I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk).... I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for this.... On isdn side I also have a problem. Asterisk quite often says that it cannot create ZAP channel, although partticular span is reported up and active..... I've also tried...
2004 Jul 04
1
How to use return value in extensions.conf
...207XXXXXXX,3,Voicemail(u${EXTEN:4}) exten => _0207XXXXXXX,4,HasNewVoicemail(${EXTEN:4}@default:INBOX|msgcount2) exten => _0207XXXXXXX,5,GotoIf($[${msgcount2}>${msgcount1}]?7:6) exten => _0207XXXXXXX,6,Send an email or something. ! exten => _0207XXXXXXX,7,Hangup However when the user hangsup the rest of the dial plan seems to be skipped. Any ideas ? suggestions. Umar.
2004 Nov 25
1
No hangup(vpb)
Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus
2005 Feb 26
1
Queue Auto fallthrough
I gave a queue setup like this, but I also have it setup so that if no agents are online, the caller cannot get in but I discovered that if that's the case, the call hangsup on the caller: [soportetecnico] ;Soporte Tecnico exten => 2,1,Playback(${SONIDOS}/transferringcall) exten => 2,2,Queue(Soporte-Tecnico) exten => 2-.,1,Playback(noagents) I want to play a message tothecaller saying no agents are online but this doesn't seem to be working... Any sugges...
2008 Aug 16
0
Getting cdr(billsec) 0 -- please help
Hi, Here is the scenario: Originating local channel using AMI. On answering the channel, it will goes to a context. Which start to playback a file. & after hangup at h extension I am caliing an agi script which insert CDR into DB. Now the problem is when I script hangsup during payback CDR(billsec) returns currect result. But when it hangsup after playback cdr(billsec) returns 0 . Please help me to find out what I am doing wrong. Thanking you in advance -- Krunal Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.d...
2006 Apr 11
1
E1 Disconnection Asterisk behind an old PBX
...users GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party decides to Hangup the call 3- Asterisks detects this disconnection and hangsup the call 1- SipUserAgent "A" Dials a call to a Mobile or PSTN phone "B" 2- "B" , the called party decides to Hangup the call 3- SipUSer Agent "A" just starts listenning the busy tone indications and Asterisk doesn't hangsup the call automatically. 4- The...
2003 Jul 22
3
busydetect and random hangups
Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress
2004 Jun 24
5
chan_capi problem - hangup???
Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10 CalledPartyNumber = <c1>...
2007 Aug 23
1
channel not hungup (zombie?) so call limit not reset to zero
im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not even hear ringing. when i use "sip show channels" command, it shows me a channel for user A like below: crunch 30d92...
2005 Jul 16
0
Hangup Detection with busydetect
My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I read on indications.conf. "diit diit diit" for busy "diit diit diit diiiiiit" for congestion...
2006 Jan 20
1
Dial command not executing following priority when caller hangs up
Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo
2008 Feb 16
0
arris tm502g cablemodem FXS ports and zaptel 1.4.8
.... Well, I used loopstart as the signal, however when using it I face one very nasty issue. My asterisk/zap channel does not detect hangups correctly. I have enabled busydetect but it's kind of unreliable. Specially when using DISA, if one of my external callers use DISA and the external caller hangsup, asteirsk wont see athing and will keep both zap channels open. I will like some suggestions with this as i am not sure if it's related to signalling in the ARRIS or maybe some tweaking i can do in the x100p (true x100p). Thanks, -- ----------------------------------------------------------...
2009 Mar 24
0
originate and local channel problem
...sions.conf: [macro-autodialer-playback] exten => s,1,Playback(${ARG1}) Originate Command would be: Channel: Local/123456 at autodialer-local Context: autodialer-local Exten: 123456 Priority: 1 Variable: PROMPT=music Timeout: 60000 Everything seems to be ok, but then B side hears prompt and hangsup, Asterisk start again execute autodialer-local context - makes a loop. Maybe someone knows how to solve this or other way to do this? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/piper...