similar to: Hangup Detection with busydetect

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

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

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

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this

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 =>

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

simple setup... two * boxen, X100P in each pots(1) --> ZAP/1 --> *(1) --> IAX2 --> *(2) --> ZAP/1 --> pots(2) with busydetect=yes and callprogress=yes in zapata.conf, calling from pots(1) through to pots(2) we get the following -- Zap/1-1 is ringing -- Zap/1-1 is ringing etc, and if nobody picks up, then it timeouts and hangups, good however, if during the ringing,

Hi All; Why busydetect=yes caused this autmatic hangup happens without any reson (while responding to entering the digits in the IVR) I do not know! And what is the solution I do not know. I used busydetect=yes and busycount=5 in zapata.conf to help me in hangup when detect the busy tone, but I faced the following problem: While I am calling to the Asterisk PBX, and during the response for the

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

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

Hi I noticed that my system has three sets of data regarding telephony behaviour in different parts of the world: 1. libtonezone , part of zaptel, and the data is from the source file zaptel/zonedata.c . Zaptel seems to use it for generating some tones. 2. /etc/asterisk/indications.conf . Asterisk uses it to play tones of busy, congestion, dialtone, etc. (PlayTones). The format is pratically

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel:

FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I

Hi everybody, I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going fine. The sound quality is excellent, there is no echo, but i have a strange problem. Sometimes calls made from SIP phones to ISDN just finished

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Aqu? tienes, [es] description = Spain ringcadence = 1500,3000 dial = 425 busy = 425/200,0/200 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 Sergio Serrano Revuelto Avanzada 7 Original Message:

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

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

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