search for: hangupcauses

I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I

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

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

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems

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

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better

...terisks running. One is a small hobbyist thing just at home, and the other is handling calls for several customers. On both, I have added the line exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE}) to all relevant contexts. On my little hobbyist box this works perfectly; all calls have their hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the production server, it only works sporadically; for 4 out of 5 calls the hangupcause field is empty. If I look at the Asterisk console, I see a message like this for every call: -- Executing [h at CallToVPBX:1] Set("DAHDI/49-1", &quot...

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

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

Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow

Ok I can't get my CDR values to set from the h extension in either 1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf ; Normally, CDR's are not closed out until after all extensions are finished ; executing. By enabling this option, the CDR will be ended before executing ; the "h" extension so that CDR values such as "end" and "billsec" may

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

Hi, Is there any way I can get exact sip status from pjsip after a dial ? or all we can get is asterisk hangup causes ? Thanks in advance. KKh

Today I get the remark that a call got disconnected after 10 minutes. This what my VERBOSE-logfile tells me : [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer via Telenet") in new stack [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:2]

I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a field called cause_code but it won't write. I belive it is because the record has already been written by the time I hit the h section of the code. How might I get this info into the CDR. I need this info for Quality of Service as well as route checking. Any ideas would be apperciated. Here is my dial line and my h

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------