similar to: Variable HANGUPCAUSE always empty with DAHDI

Displaying 20 results from an estimated 2000 matches similar to: "Variable HANGUPCAUSE always empty with DAHDI"

2011 Jan 17
0
Sangoma A104d / overlapdial=yes / dial with audio one-way issue
Hi, I am facing an audio-problem with the dial application and I (!) think, that it is connected to the dahdi parameter "overlapdial=yes". Sangoma support does not see any connection between this. But when enabling this option I face with some(!) dial-partners a audio one-way issue (the called party can not be heard). Only using PSTNS (germany e1 trunk) - no voip. Is there any
2006 Apr 21
0
HANGUPCAUSE on SIP channels
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
2018 Mar 14
2
DIALSTATUS vs HANGUPCAUSE
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
2015 Oct 07
2
Storing HANGUPCAUSE in CDR
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
2018 Mar 14
2
DIALSTATUS vs HANGUPCAUSE
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
2010 Dec 22
0
Include ${HANGUPCAUSE} in CDR
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
2009 May 18
0
${HANGUPCAUSE} is not printed when call ends or is interrupted
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]
2006 Apr 13
0
Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk
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
2015 Oct 09
2
Storing HANGUPCAUSE in CDR
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
2009 Jun 29
0
FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan
Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???
2006 Apr 04
5
Hangupcause is not enough on PRI
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
2015 Mar 25
0
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit200 at gmail.com> wrote: > hello list, > > i have asterisk 11.15.0 and i have some trunks sip from my provider > > we have some ip phone astra 6731i > > each Ip-phone is configured with trunk and we call > > no ihave configured another trunk from the same provider in my asterisk > > i can call
2015 Feb 22
2
dialplan contexts syntax and terminology
I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk
2009 Mar 15
1
X-Asterisk-HangupCause - how to disable this?
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
2006 Nov 08
1
HANGUPCAUSE for unalocated number?
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.
2008 Nov 26
0
CDR Hangupcause
Hi, I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this option: endbeforehexten=yes is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set CDR value with that value. It seems to finish the CDR record before h is executed. I'm using cdr_mysql. Any idea?? Thanks!! -------------- next part -------------- An HTML
2019 Nov 16
2
Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer
What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones? I do not know if the bug is in Android native SIP, or Samsung’s build of the SIP client, or if the bug is even with the OpenVPN client, or where the bug actually is. The ACK might even be sent for real, but have the incorrect source IP so
2012 Mar 08
1
Using the h and DeadAGI
Hi All; Really I need to know why when using the "h" in the exten =>, then we use DeaAGI with it? I am using vicidial and I see this line alot, so I need to know how it work (when it will be executed): exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) The question is: When
2015 Feb 22
0
dialplan contexts syntax and terminology
READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22,
2007 Oct 25
3
Getting SIP Response Code from HANGUPCAUSE
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