similar to: HANGUPCAUSE on SIP channels

This is the first project I'm taking an active interest in, so I trust that the guiding hand of flamethrowing will teach me the ways. I was anxious to start playing with Vorbis somehow, so I downloaded it via CVS and tried to get the tool in vorbis/cmdline to compile. It didn't compile at all until I made the attached changes. Note that I'm only trying to guess as to how the

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

Hi all, I have seen a lot of message talking about asterisk crashed when using queue and mixmonitor together. I do use both in our system and also get the crash (segfault) randomly. I don't know it is related to the reason above as I have no idea about how it happened. I get the core dump below. If anybody has any idea about the root cause of the crash, please tell me. Asterisk 1.4.13

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I'm getting a new 1.6 box built out and working, and wanted to emulate the functionality of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86

Hi there, I am attempting to set up a simple BRI and SIP based platform using * with the quadbri card (it's not sharing an IRQ). I enclose my zaptel and zapata.conf files. For the inital test I'm simply trying to connect to the * demo menu. The drivers compile (with a few warning that I believe aren't important - see attachments). chan_zap comiles with the warning: chan_zap.c:

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

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

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 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

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]

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.

** THIS IS NOT WHERE YOUR REPLY BELONGS ** On Wednesday 25 Mar 2015, Salaheddine Elharit wrote: > tnaks for your response but the number dialed exist and i can call this > number when i configure the trunk directly in x-lite and i call call also > this number from my cell phone . > any help > thanks and regards Make sure you are sending the number in the correct format, when you

Hi List We want to be able to reject some pjsip calls with a 'temporary' failure so that the PBX of the caller plays an announcement in the language of the caller, that the call can temporarily not reach the destination. https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings Case 41 seems right... Haugup(41) causes the call to be rejected with '603' causing the

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

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