Asterisk Hangup Extension

Hangup handlers. I have the following macro in my diaplan which is excuted each time an incoming call comes. h extension to execute after a hangup. asterisk sms (re)delivery: gistfile1. At least until others show up. The Asterisk binding is used to enable communication between openhab and the free and open source PBX solution Asterisk. 2 support it). Extension hangup-handlers,s,1 have be valid and exist. A nice graceful exit in the dialplan, I indicate by playing a sound. While not strictly necessary due to auto-fallthrough (see the note on Priority numbers above), in general we recommend you add the Hangup() application as the last priority in any extension. php user login. x can be set up using arr or ars. Now dial extension 2000 with your phone. If doing a soft hangup on them doesn’t work, the only other way I know to do it is to restart Asterisk. The protocol has the following characteristics: By default, AMI is available on TCP port 5038. asterisk dial hangup direction - Is there a variable I can use to record the direction of hangup? If CALLER hangup you will go 'h' extension in same context, but. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the callee to extension 123. When hangup handlers are executed Any hangup handlers associated with a channel are always executed when the channel is hung up. Call and hangup using Asterisk as a SIP client exten => _XXXXXXXXXXXXXX,2,Hangup it's rather convenient to have a softphone hooked up as an extension to test. Sugar Asterisk CTI integration integration improves efficiency of your phone communication by giving you more information and more options for each call you make or receive. Asterisk (PJSIP) pjsip. when you transfer the calls, asterisk will search for the extension in your current context so if someone calls using "sales" he will be able to transfer only to extensions 41XX, if you want to let him transfer to extensions 40XX then you should add 40XX to sales context, example:. Add Extension User Extension. Here is my revision of RonR's method - this uses Asterisk's Bridge application, rather than the Asterisk Parking Lot. The configuration for the. Click To Call Chrome Extension provides click to call facility from any web pages of Chrome Browser by selecting number from web page. 2 and Asterisk 15. Maintenant, vous pouvez enregistrer votre fichier extensions. Why You Should Hang Up Immediately When You Get a Robocall. I wanted to do a little in AEL just to get a feel for it. conf queues. SIG is one of many extensions to Q. Same thing works fine in Asterisk 11. Join GitHub today. I also have problems with Asterisk. If extension 3030 is dialed, Asterisk attempts to connect to the given channel cisphone001, using SIP. Button 1: *401. conf and sip. Asterisk installations of the sort I had done previ-ously, where Asterisk functioned as an answering machine or a small office phone system, have been documented in many places. 4 with asterisk 13 LTS and I think its been properly integrated. Here's the scenario: Mark picks up his phone (1000) and dials Richard by dialing 2000. (Later we will look at things like voicemail. Here is my revision of RonR's method - this uses Asterisk's Bridge application, rather than the Asterisk Parking Lot. When a call is made to extension 123, Asterisk will answer the call itself, play a sound file called "welcome", give the user an opportunity to leave a voicemail message for mailbox 44, and then hang-up. Install and configure IVR in Asterisk Server. conf file which is located in /etc/asterisk/sip. Asterisk CLIThe Asterisk command line interface (CLI) is reached by usingthe Linux shell command asterisk -r If you want debugging output, add one or many v:s asterisk -vvvvvr The Asterisk server has to be running in the background for the CLI to start. ms:5060 ; (one of our multiple servers, you can choose the one closer to. The real problem is that he didn't terminate the call, so in the absence of that Asterisk reverts to a hangup, and in Asterisk that is implemented by answering the call for a fraction of a second (if it is not already answered) and then hanging up (I have no idea why). Just an FYI, in the dialplan below, The ReceiveFax() application receives the fax document and then automatically hangs up the call when it is finished. Asterisk has nearly two hundred included applications. 2 and freepbx with asterisk 1. Well clearly this isn't really secure way of doing things. Hangup handlers can also be attached to any call leg because of pre-dial routines. Asterisk has nearly two hundred included applications. With the end of maintenance support after the releases of Asterisk 1. When you hang up the phone, the channel is deleted and any variables in that channel are deleted as well. You won't find here instructions on setting them up here. In combination with other bindings (e. Here is the situation: I have 2 extensions on 2 phones. WebRTC / Asterisk 11 / FreePBX testing Raspberry Pi 2 WebRTC and websockets support for Asterisk and Freepbx. The extensions. The extension includes a list of dialplan applications which will be executed on the channel. First, Asterisk needs to pick up the phone and compare the current local time to the 8 am to 10 pm window. More than 3 years have passed since last update. Our extension 1001 has three priorities: exten => 1001,1,Answer() exten => 1001,2,Playback(hello-world) exten => 1001,3,Hangup(). Some mobile systems and some VoIP systems treat CLEAR as hold and REANSWER as unhold, rather than responding to CLEAR with RELEASE. Great so now Asterisk is able to decide which Voicemail message to play based on whether or not we are on the phone. Send calls to the entered number and connect it to an extension 1000 in the system. Asterisk installations of the sort I had done previ-ously, where Asterisk functioned as an answering machine or a small office phone system, have been documented in many places. conf type stuff. When the conversation is over the Hangup application will be executed and the Asterisk PBX will hang up the line. This will call forward extensions 101 to extension 610. I decided to write a book and it was published in 2005, named "Configuration Guide for Asterisk PBX", translated to Portuguese and Spanish. I saw the asterisk console telling me recording were being played, but no sound. When the A side hangs up, or when the B side hangs up and no Dial, etc. The Asterisk binding is used to enable communication between openhab and the free and open source PBX solution Asterisk. Because Asterisk runs on Linux you can leverage existing tools to help interface and manage Asterisk. FOP2 is a web based switchboard for the open source projects Asterisk© and FreeSWITCH©. If you don't have a hardware SIP phone, you can use a software SIP phone that you install on a client computer. It is assumed you already have Linux and Asterisk and FreePBX installed using a procedure similar to this one. I have also made an assumption that you know how to install asterisk and configure SIP Peers/Trunks. sample) #touch extensions. Im wondering becuase by default it doesnt. this is the context of extensions_custom. Developed by Mark Spencer. , extension 153 will cause the SIP telephone set on John's desk to ring), in an Asterisk dialplan, they can be used for much more. (1) The system works fine when I use the default voicemail app. In this case we used extension 1111200, which will be auto answered by the Asterisk server, and recorded. Simple Asterisk VoIP on a hosted server I’ve been playing with Asterisk for a long time, mainly as a hobby and mostly just hacking things together. Asteriskで使用するダイヤルプランの構文について解説しています。 exten => 201,2,Hangup() その際「extensions. restart when convenient - Restart Asterisk at empty call volume sla show - Show status of Shared Line Appearances soft hangup - Request a hangup on a given channel stop gracefully - Gracefully shut down Asterisk stop now - Shut down Asterisk immediately stop when convenient - Shut down Asterisk at empty call volume stun debug - Enable STUN. org runs on a server provided by Digium, Inc. , the Samsung TV Binding) you can display caller IDs on your TV. Asterisk SIP configuration is done is sip. The system supports eight lines, 24 extensions, and over 36 voicemail users. Star codes are known to be an easy way to enable or disable certain features in many Asterisk/IP systems. 2, see note below if using Asterisk 1. Here's an extensions. exten => h,1,Hangup(). When a call comes in from the PSTN, however, Asterisk doesn't know what was dialed or whom the caller is trying to reach. Learn how to configure an Asterisk SIP extension on Ubuntu Linux version 16, by following this simple step-by-step tutorial, you will be able to create a basic SIP extension using the Asterisk server. Grygorii Maistrenko homepage. You can't share subroutines between different logics if the hangup are different. One for your phone and the other for you laptop and everyone in the office has a similar configuration. 8 g729 for all calls. Extension states are another important concept in Asterisk. Restart asterisk ; Note: asterisk has to be started "after" festival. We recommend using 3- or 4-digit extension numbers. It can be used for calling via the landline but also with appropriate hardware using VoIP. All of the subsequent priorities would have to be manually renumbered. IAX2, ISDN, and SS7 are all subsets of the cause codes listed above. Hello Experts, I have setup a asterisk 1. 4 dialplan I have a. Some devices use this to auto-program the voicemail button on the endpoint. (2) Even when I use app_voicemail_imap, the system seems to be detecting the hangup -- there's an entry in the logs just like when I'm using the default app. What is that, you say? DISA is Direct Inward System Access, which on a PABX (of which Asterisk is a fine example) means getting a dial-tone like you are a local extension. Also note that this dialplan is designed for four digits extension scheme. It is a little AEL and a little regular extensions. sh file is chmodded and asterisk/cme link is working good. Installing The Asterisk PBX And The Asterisk Web-Based Provisioning GUI On Linux. A nice graceful exit in the dialplan, I indicate by playing a sound. Display Name. In this case, you should take care how to use asterisk hangup handlers into subroutines. Unfortunately, if there is a hangup extension, the bridge CDR is actually swapped out with the channel CDR. How to setup Vtiger CRM Telephony Integration with Asterisk August 3, 2016 Smackcoders Human has been answering telephone for a long time without knowing who is on the other side. Do you have a solution? I am using one (exten => h) for the whole dialplan, so it's hard to assign a new variable in each dialplan extension to handle the called number. Add Extension User Extension. Creado por Wiki Bot, modificado por última vez en dic 26, 2013; hangup all calls on extension 140. Asterisk is a very powerful media server for call routing and with great design and configuration can be used sustainably in a company,institution or office. The hangup cause AST_CAUSE_NOT_DEFINED is not actually a Q. SIP Configuration. Problem is that it is repeating itself indefinitely. My router (a Draytek) is placed into the DMZ of my internet (100 Mb/s) modem (192. A pc with linux and asterisk installed on it. 200 nolu dahili tanımı olmadığına emin olun. Analog will always have a hangup cause code of AST_CAUSE_NORMAL_CLEARING. Asterisk SIP configuration is done is sip. We recommend using 3- or 4-digit extension numbers. Download Presentation Asterisk Jargon An Image/Link below is provided (as is) to download presentation. 4 with latest update ( >10 callcenters for >6 month). Here is the situation: I have 2 extensions on 2 phones. The real problem is that he didn't terminate the call, so in the absence of that Asterisk reverts to a hangup, and in Asterisk that is implemented by answering the call for a fraction of a second (if it is not already answered) and then hanging up (I have no idea why). Now let's put Answer(), Playback(), and Hangup() together to play a sample sound file. Agent can hang up the call by clicking the hang up button. You'll need your SIP-ID and SIP Password. conf extensions. It's just another "extension" to Asterisk. And there's no limit to the number of call queues on your system. conf examples. Voicemail Extension. From the drop down click Extensions; Adding a Virtual Extension. 14 I’m using CENTOS 4. For each endpoint that SIP Server monitors, configure a hint entry to ensure that Asterisk will accept a presence subscription (from SIP Server, in this case) for those endpoints—for example: exten => 2001,hint,SIP/2001. Configure an extension on the Asterisk server to be recorded. Here is an example which answers the call and reads back any digits you press. Extension hangup-handlers,s,1 have be valid and exist. 0 Setting up Voice Mail in Asterisk 11. Join GitHub today. Protocol Overview. FreeSWITCH Side. Secara umum, setiap ekstensi dalam Asterisk merujuk pada user tertentu yang ter-register ke Asterisk tersebut sehingga biasanya nomor ekstensi sama dengan id user. txt) or view presentation slides online. Venezuela and Spain Asterisk Dialplan Posted by silvinux ⋅ October 23, 2014 ⋅ Leave a comment Filed Under Asterisk , dialplan , dialplan españa , dialplan spain , dialplan venezuela , extensions. - Support Forums and Community › Miscellaneous › How-To. ms:5060 ; (one of our multiple servers, you can choose the one closer to your location) server_uri = sip:atlanta. For outbound calls from Asterisk PBX to GoTrunk SIP Credentials (SIP username and password) authentication is used. How to save a message for IVR in asterisk voip. As a valued partner and proud supporter of MetaCPAN, StickerYou is happy to offer a 10% discount on all Custom Stickers, Business Labels, Roll Labels, Vinyl Lettering or Custom Decals. Asterisk CLI provides Hangup command to hangup live calls. Configuracion de un IVR en Asterisk - Segunda parte Mi, 14/01/2009 - 08:45 admin. so decide which once you want and download the source file ** Asterisk 1. Now dial extension 2000 with your phone. Hello, I have a question regarding incoming fax to local file (on the Asterisk server). The AGI (Asterisk Gateway Interface) facility allows you to launch scripts, from the Asterisk dial plan. In combination with other bindings (e. I have tried the following code to dial a sip extension and follow me number if the extension is not answered. The state of an extension is determined by checking the state of one or more devices. Asterisk is a great opportunity for thousands of developers, resellers, system integrators, ITSPs, contact centers and small to large companies. ulaw) same => n,Dial(SIP/101) In another example if you want to record call on user extension 101. 2系ではextensions. You should hear the default music. From the drop down click Extensions; Adding a Virtual Extension. However, we can "prolong" the life of the call past the hangup condition with a special extension, "h". The various other methods are wrappers for FastAGI calls, which execute on the Asterisk server. Now dial extension 2000 with your phone. x can be set up using arr or ars. Once the questions are done, the caller is hungup on. Both use the special i code, meaning an invalid — unhandled — extension was entered. There are only minor differences between the Asterisk dial plan code and the Dial Plan Compiler code; mainly, that you only type the extension number once, including its context, and below it you put all your dial plan code, without line numbers and without repeating the extension in every line. More specifically, for every extension I want to know: If the extension is in a call, what is the unique ID of that call, what is the caller id, what phone number was called (incoming line). # Binding Configuration. Giving the ability for Lync users to call dial Asterisk users directly using 1xxx Giving the ability for Lync users to make inbound and outbound calls using Asterisk as a back to back user agent. His softphone appears to be "in use" (for about 30 mins) but it is not actually being used - as confirmed by the guy sitting nearby. Same thing works fine in Asterisk 11. ; ; Static extension configuration file, used by ; the pbx_config module. Servidor Asterisk (Instalacion y Configuracion) 1. The extension includes a list of dialplan applications which will be executed on the channel. Could you please give me advice ? Here are my extensions. Asterisk Dictate and the old Hangup Issue I implemented a custom phone based dictation solution using Asterisk PBX and the Dictate app and noticed that anything in the dictation dial-plan after the Dictate command, was never executed *if* the call was dropped or hung-up by the caller. [email protected] [Index of Archives] [Asterisk Announcements] [Asterisk App Development] [Gnu Gatekeeper] [IETF Sipping] [Fedora Linux Users] [Yosemite News] [Deep Creek Hot Springs] [Yosemite Campsites] [ISDN Cause Codes] [Asterisk Books]. Secure Your System. #cd /etc/asterisk (this is the directory where the asterisk configuration files are) #mv extensions. Lets have a look an example: sip. After the call is answered, you can press *9 to add the dialed number to the blacklist. INCOMPATIBLE_DESTINATION: The call was unable to be connected to the other party; typically caused by incompatible codecs between the respective devices. * AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers * AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR 2012-07-30 Asterisk Development Team * Asterisk 10. Important: To log stuff to the console, either use Verbose(), or use NoOp() but the latter will only work if you set "verbosity" to at least 3 (in the console, type "set verbose 3"). I inherited an Asterisk 1. Button 2: *402. It may also be that this one worked perfectly with previous versions of Asterisk, and isn't working with the version I'm trying (this could be more probable: I've seen people reporting many broken things in 1. The Asterisk dialplan's configuration file name is extensions. conf (extension dialing and Vo-IP inbound/outbound). Definition at line 56 of file channel. Asterisk 1. The next step is to install the ngSMS extension to Asterisk PBX. exten => _*402,n,Hangup() Now, on the phone, I create a speed dial button for each. Asterisk is software that turns an ordinary computer into a voice communications server. In our example, you can dial the extension 100, after the beep you can start recording your message. This means that if somebody calls you and you are either talking to somebody else or you are otherwise not available then Asterisk can answer on your behalf and give the calling party the possibility to leave a message for you. If you called the video-conference extension (If you didn’t try now. Jetzt hänge ich an dem Punkt die erste "test" extension für das interne isdn zu erstellen. Following is my extension. Don’t worry! Take a deep breath and keep reading! This is for Vanilla Asterisk 1. Each phone is configured as an extension in the PBX but the greatest advantage of Asterisk is that the extension does not have to be in the same physical location. Extension states are another important concept in Asterisk. For any scenario in which we cannot determine the number dialed, we use the s extension. Some devices use this to auto-program the voicemail button on the endpoint. 2, possibly a bug. esta es la tercera línea. " actions · 2014-Apr-28 12:23 pm · rsriram22. ms:5060 [voipms] canreinvite=no context=mycontext host=atlanta. 2 is Asterisk with extensions in the range 2000-2019. By default, the file /etc/asterisk/sip. And finally, you can think of the asterisk in the ISN as the ‘at’ symbol (@) in the ISN. net projesini. The current version (CVS date 19 Jan 2004) is a single executable file. And there's no limit to the number of call queues on your system. It concentrates on the PBX in a Flash distribution using FreePBX as the web based administration tool. The hangup cause AST_CAUSE_NOT_DEFINED is not actually a Q. conf file? Cause i try to make: exten => 7777,1,Dial(a2billing,${EXTEN},1) But asterisk is not finding that extensions, and it’s not working, how can i create that extensions in asterisk files directly without freepbx? Thanks in advance. WM, I’m an asterisk user, and not sure which version [email protected] is based on, but in the current version of asterisk, you can make life a lot easier by using "n" instead of a priority number (except for the first priority in an extension, which must be 1). Now dial extension 2000 with your phone. Asterisk Tutorial Presentation V1. Display Name. If doing a soft hangup on them doesn’t work, the only other way I know to do it is to restart Asterisk. iaxComm is an Open Source softphone for the Asterisk PBX. Asterisk© and FreeSWITCH© are powerful and complex softwares. Configuracion de un IVR en Asterisk - Segunda parte Mi, 14/01/2009 - 08:45 admin. Follow this tutorial to get working IVR on Asterisk and test it. Once the questions are done, the caller is hungup on. Then enter the channel number you want to close then press enter. 4) If key 2 is not flashing, pressing Key 2 will initiate a call to JaneDo1 at extension 502 Asterisk Configuration: In extensions. With the manager interface, you can control the UCx to: originate calls, check mailbox status, monitor channels, queues and also execute commands. The extension making the call does not have appropriate toll access to complete the call. Analog will always have a hangup cause code of AST_CAUSE_NORMAL_CLEARING. Only change this on devices that have special needs. Asterisk is an open source/free software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Como cualquier PBX, se puede conectar un número determinado de teléfonos para hacer llamadas entre sí e incluso conectar a un proveedor de VoIP o bien a una RDSI tanto básicos como primarios. conf examples. SIG is one of many extensions to Q. I also have problems with Asterisk. We need to route calls made on freeswitch to the 2000-2019 extensions to the asterisk box, we'll use our external sip profile for this but internal should work, as well. You can chose an […]. (Turns out I forgot to tell asterisk to activate NAT on the extension). in the extensions file, by issuing the following command. The protocol has the following characteristics: By default, AMI is available on TCP port 5038. After a successful reload, we run the dial command which starts executing the extension helloworld located in default context. We use your LinkedIn profile and activity data to personalize ads and to show you more relevant ads. phone - sip phone number sip. Same => n,Hangup. For example you want to record the calls coming on DID 1949 555 55555 exten => 19495555555,1,MixMonitor(${UNIQUEID}. sh file is chmodded and asterisk/cme link is working good. Asterisk© and FreeSWITCH© are powerful and complex softwares. I have the following macro in my diaplan which is excuted each time an incoming call comes. #Asterisk PBX Basics Asterisk turns an ordinary computer into a communications server. I am experiencing weird problem in Asterisk 16. After the "beep", hangup. Hanging up now. As a result, priorities 11 and 12 are not reached. For inbound calls to one of Telephone Numbers on your GoTrunk account to work Asterisk PBX needs to Register with GoTrunk service (and periodically refresh registration in case IP address changes). Extension hangup-handlers,s,1 have be valid and exist. If you record all the calls directly to the HDD in asterisk pbx and you got a large call volume (number of calls) then it will damage your PBX’s HDD very soon. conf and sip. Login to your asterisk CLI console. conf (domain name, Vo-IP registrar, users who can connect directly) extensions. ASTERISK - Asterisk adalah sebuah software hybrid TDM dan PBX packet-voice yang memiliki platform IVR dan ACD dengan kode sumber terbuka - Asterisk berlisensi GPL dan non-GPL dan ditulis dengan C. If the extension will be used from a phone behind a NAT-based router, change the NAT entry to Force,Comedia. Following is my extension. Select this checkbox, and you will be able to connect a different PBX to that extension. Asterisk picks up the line, plays the hello-world sound file (which is installed with Asterisk) and hangs up. This is an efficiency measure in Asterisk, as it's much less computationally. ) Now lets update our extensions. conf file is the most important file, and will be discussed later. , please call John at extension 153), they can be used for much more in Asterisk. The Asterisk Manager Interface (AMI) is a monitoring and management interface over TCP. That will bring up “soft hangup” again. GitHub is home to over 40 million developers working together to host and review code, manage projects, and build software together. Modify yours accordingly, esp XXXX part … Note the pickupchan module worked on asterisk 1. Put them in extensions_custom. extensions. Phone number is the customer number. Sometimes it is necessary to kill unwanted phone calls, or just to free up the system from a call which is in a hung state: it's marked as active, but there is no call there anymore. This means that if somebody calls you and you are either talking to somebody else or you are otherwise not available then Asterisk can answer on your behalf and give the calling party the possibility to leave a message for you. , extension 153 will cause the SIP telephone set on John's desk to ring), in an Asterisk dialplan, they can be used for much more. A nice graceful exit in the dialplan, I indicate by playing a sound. After all this modification which we made, the unique way to solve it for me it was to turn account for multiple access to avoid to be blocked when in use still showing more then 0. 今日は、昨日の天気とうって変わって、朝は、霧雨が降っていました。 昨日の動物園で撮った動物たちの面白い光景がありましたので、合成写真にしてみました。. Well clearly this isn't really secure way of doing things. Asterisk Dictate and the old Hangup Issue I implemented a custom phone based dictation solution using Asterisk PBX and the Dictate app and noticed that anything in the dictation dial-plan after the Dictate command, was never executed *if* the call was dropped or hung-up by the caller. However, asterisk doesn't seem to detect when the other party on the call has hung up the connection. conf and sip. conf is the most important Asterisk file and it has the main objective of defining the PBX dialplan for each context and therefore for each user. I have read about Asterisk and wanted to test it out as I will be managing/troubleshooting it at work anytime soon, so I thought of getting my hands dirty and getting some basic experience on it. Asterisk & ENUM Extending the Open Source PBX Michael Haberler, IPA Otmar Lendl, nic. Asterisk; News Archives (older news). exten => 1001,n,Hangup;Logoff this way works but is not very intuitive because you;have to hit the # key when prompted for a dial back extension;it really doesn’t make sense to endusers; exten => 1002,1,AgentcallbackLogin(${CALLERID(num)}||) exten => 1002,n,hangup [savelono-queue-out] include => from-internal. File: /etc/asterisk/extensions. Both use the special i code, meaning an invalid — unhandled — extension was entered. 0 Asterisk 1. From the drop down click Extensions; Adding a Virtual Extension. When a call comes in from the PSTN, however, Asterisk doesn't know what was dialed or whom the caller is trying to reach. For example, if a Dial to a SIP UA is cancelled by Asterisk, the SIP UA may not have returned any final responses to Asterisk. You have use. At this point the channel is hungup and you should only be gathering information about the call for further processing later. Hangup handlers can also be attached to any call leg because of pre-dial routines. Well I had to surf a lot to find the exact command to hangup calls in the latest Free PBX so here I will show you in easy steps for how to hangup active calls on PBX. Same thing works fine in Asterisk 11. Asterisk juga memungkinkan komunikasi antar pengguna telepon regular dengan telepon berbasis sip (sip phones). A detected parking extension converts the blind transfer into a pseudo attended transfer that completes the transfer after the announcement. conf file which is located in /etc/asterisk/sip. c: RTP Read error: Connection reset by peer. For example, if a Dial to a SIP UA is cancelled by Asterisk, the SIP UA may not have returned any final responses to Asterisk. (Turns out I forgot to tell asterisk to activate NAT on the extension). The patch process is simple enough: SSH into the FreePBX box as whatever user you use for this purpose, I'll select root for this document. Extension states are another important concept in Asterisk. Installing The Asterisk PBX And The Asterisk Web-Based Provisioning GUI On Linux. However, asterisk doesn't seem to detect when the other party on the call has hung up the connection. The channel technology specific hangup cause information; A text description of the Asterisk specific hangup cause; Note that in some cases, the hangup causes returned may not be reflected in Hangup Cause Mappings. (3) What's missing in the logs is the entry , which appears when. In this example we are going to create two new users with their extensions and their voicemails. 1 - Install Centos 4.