asterisk show active calls duration. Hence, receiving no results at all could be a red flag the number is spoofed. Select the Advanced tab, and then look for yellow and red items which indicate poor call quality or connection problems. uart_dump call_state span_state at_parse at_handle sms_encode sms_decode. watch "asterisk -vvvvvrx 'core show channels verbose'" Watch active channels in Asterisk 1. You can use different search criteria. In the stress and load testing I’ve done, I’ve not seen Asterisk break a sweat. This is how FreePBX starts asterisk and any other processes it need. Use the 100 extension to call 666 and enter the PIN 5555 to create a conference bridge. cynjut (Dave Burgess) June 25, 2019 . The call description is as received in the dispatch center; the final outcome of the investigation may be different. js --gulp --dry-run to do a one time analysis. asterisk console commands · GitHub. Watch number of active channels. If you have many actives calls, there's no way to filter all the output in the CLI, which makes the CLI pretty hard to read for specific channel(s) and not to all channels. The incoming codec can, theoretically, change from frame to frame, when using VoIP type channels. 2: “channel request hangup ” show voicemail users: List defined voicemail boxes. Our strategy is UC, and sipXecs is a native SIP solution consistent with our strategy. conf there is an option for every peer called qualify. 2) I have a server application which will need to connect to asterisk and retrieve all the channels already active. Simple command is to enable SIP debugging for one phone with: SIP SET DEBUG PEER PEERNAME. All information is taken from Mysql. 38 negotiation requests will be accepted and relayed. 2, Asterisk addressed this problem: it introduced the use of the n priority, which stands for “next. Step 2: Registering the account on the Sip Server. Obviously, the more information you have about the call, the faster you will be able to pin it down. Asterisk Feature Busy Lamp Field (BLF). Use the 200 extension to call 777 and enter the PIN 1234 to join the conference call. asterisk2*CLI> core show channels Channel Location State Application (Data) SIP/3224-00000a19 [email protected]:42 Up Dial (SIP/4027,15,trI. We can also pause and unpause members in a. A Long Term Support release is fully supported for 4 years, with an additional year of maintenance for security fixes. phone call from dial to hang-up in seconds (integer) Duration of phone call . Login to your asterisk CLI console. How to Use the Netstat Command. conf Welcome ASTassistant is easy to use freeware for managing phone calls via the Open Source Asterisk PBX platform. The development team is committed to keeping the content up to date and accurate. Configuration file for Chan_DAHDi lookups in Asterisk!Sample chan_dahdi. A make a phone call to 12345678, and H pick up the phone call; then A tell H that he want to contact the customer inside Room100, after authentication, H TRANSFER THE PHONE CALL TO B AND HANGUP. 107 E-model which predicts quality on MOS scale. That number is an indication of the calls processed since the service is running or last asterisk. The asterisk manager API (AMI) can be used to monitor calls realtime. exten => 220,1,Dial (SIP/220,30,m (newclass)) After changing the configuration, restart asterisk: 1. They'll also be able to see if you're currently in the. Restart asterisk: asterisk -rx "logger reload" Restart syslog: service syslogd restart (or rsyslogd) Manual data import. how can i show Aterisk active call dutation only call time?. Some reporters said it was caused by call pickup *8. SIP Registry - How many SIP connections Asterisk is registered to. Make sure the existing view refers to the correct BOM. In this case, the command is useful in order to display dial peers and capabilities associated to an active call. js + nami what the best way to get this information? tried use an action. This is a very basic call flow showing an incoming call to the PBX that is routed to an IP phone. console send text — Send text to the remote device. to dial to park: parkext => 700 ; What extensions to park calls on: parkpos => 701-720 ; Which context parked calls are in, ; need to INCLUDE this in extensions. In order to find the hex ID, enter the show call active voice brief command or use the show voice call status command. If you have not much resources and experience, that will be just wast of time (alot of things can go not as you expect). When Mark picks up his phone, Asterisk will dial extension 2000 for him. Let’s say my server crashed and i have several calls in progress. console {mute|unmute} [toggle] — Disable/Enable mic input. it appears on my phones, however in a missed call email with the above script, instead of the number and name, I get two copies of the number. sudo asterisk -x 'core show channels concise' | awk -F! '{ if(NR>1) print $1 " ("strftime("%Y-%m-%d %H:%M:%S", $12)")" }'. “Polling Strategy” – define how calls are distributed to agents: Hunt random start – randomly choose an agent to distribute the call to and evenly distribute. It’s just doesn’t ring at the other side. For see currently active calls you can use ForkCDR functions of CEL events (like some commercial billing systems use). I am using AMI, but seems like there is no method to get answered time or billsecs of all active calls. [quote="david55"]core show channel will give you the contents of the CDR answer time field. While troubleshooting Asterisk phone system I figured that I needed a way to see how to show Asterisk calls in progress. watch "asterisk -vvvvvrx 'core show channels' | grep calls" Watch active channels. view-active-calls-details-on-freepbx. Duration of each call was random from 30 to 40 seconds. If you want to use “Contacts” then, as suggested, use the Contact Manager. It collects metrics by polling the Asterisk Manager API remotely using an HTTP agent and JS preprocessing. ConfBridge is a high definition-capable conference bridge component that makes it easy to build stand-alone conferencing services or to integrate conferencing into other solutions, including IP PBX systems. Channel Context Extension Prio . Total Duration = invite time + ringing time + call time. xxx blocked by fail2ban: iptables -D fail2ban-asterisk -s xxx. Mark then hangs up, picks up the phone and dials 30 to call the CallCompletionRequest application. The “Asterisk Phone Book” is not contact info. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G. In a nutshell, it consists of a list of instructions or steps that Asterisk will follow. Asterisk call statistics dashboard for Grafana. Select Call history, and then select the call or meeting that you want to troubleshoot. Troubleshooting and Debugging VoIP Call Basics. —Albert Einstein (1879–1955) The dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. community and would have not been possible without your participation. Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg. We installed Asterisk PBX using Ubuntu software center, and used default configuration. Asterisk provides specific support for the management of phone calls received by an incoming call center. By default it is /var/lib/asterisk/mohmp3. That is, a phone, a PBX, another Asterisk system, or even Asterisk itself (in the case of a local channel ). Make HTTP request (API call: Post & Get) to the external system to fetch data during call session (Active Channel) Integrate Asterisk to MySQL Database, MSSQL, or any database of your choice Basic Troubleshooting in Asterisk PBX Secure your asterisk PBX Requirements Asterisk Knowledge: Yes (Not a show-stopper with focus and follow the course as. Call pickup allows you to answer a call while it is ringing another phone or group of phones (other than the phone you are sitting at). Requesting to pickup a call is done by two basic methods. It is calculated against a table called "Time1secIntervals" withe column "IntervalStart" holding the 1 second time intervals. Asterisk includes a standard application called ConfBridge. The dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. c:1162 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/[email protected];1 When this happens the system will not accept any more calls and in order to make it work. Check current active calls. How to get Originate call duration in Asterisk?. 2 uses the following:-> pri set. The type of release defines how long it will be supported. Optional: To block IP address xxx. Output needed data to file in manual mode. In certain situations, the DAHDI requires a restart to apply changes to any tdm-config, tdm-profile, fxo-profile, or fso-profile configuration parameter that the system cannot update by simply reloading the configuration. We would like to know the CLI command (or AMI) to display the audio codec which is being used in an ongoing call, please. Dialplan information is located in several conf files (please check official Asterisk docs on this). I’m banging my head against the wall now as I don’t really understand what I’m doing wrong here. This commit does not belong to any branch on this repository, and may belong to a fork outside of the repository. I’ve connected the OPEN 79XX telephone database to my asterisk/freepbx setup. Please note this does not mean active calls, as a single call can be 2 or more IAX2 channels. console hangup — Hangup a call on the console. Run crontab (under root) and create a task: sudo crontab -e. To remove Call Forwarding manually: Press Feature 11. Next do an Asterisk reload to tell Asterisk about the new extension. i tried "core show channels verbose" this command to get output about active call. Get detailed channel information If you want to see a detailed information on the channels in asterisk, use the command. Response: response by Asterisk to the client action. This switch displays active TCP connections, TCP. Exceptionally long voice > queue length queuing. The output of the command asterisk -vvvvvrx 'core show channels verbose' shows total number of calls processed. echo canceller on the channel (if any), for the current call only. In our case this will cause the dialing of the user operator through the IAX2 channel. Replicate the issue, then download the full Asterisk log located at /var/log/asterisk/full, and send to Telos Support along with information that can be used to identify the issue, such as:. An overview of the Voice Call Flow and Telephony Architecture in a Cisco Router is presented, followed by a step-by-step VoIP troubleshooting approach presented in these steps: Verify digital and analog signaling. A channel is an entity inside Asterisk that acts as a channel of communication between Asterisk and another device. You will see this prompt after the basic Asterisk information is displayed: asterisk*CLI> To change the verbosity of the console, use the following: core set verbose 4. conf: context => parkedcalls ; Number of seconds a call can be parked ; for (default is 45. The main idea is to filter per channel(s) via active calls or regex to map your channel name(s). Suppose that Asterisk is already installed. This application will place calls to one or more specified channels. Motion-PBX*CLI> sip set debug peer giove1motion SIP Debugging Enabled for IP: 151. By default, Asterisk uses Dialplan to route the calls to various other places. dicko (dicko) June 25, 2019, 6:13pm #8. Watch active calls on an Asterisk PBX watch -n 1 "sudo asterisk -vvvvvrx 'core show channels' | grep call" Show active calls as the happen on an Asterisk server. Alarm in seconds for call duration in the realtime view. * send a second request to originate your call. But next time we restarted asterisk the registration kept on timing out. console transfer — Transfer a call to a different extension. This information is delayed by approximately 20 minutes. Here is the screenshot for your reference. For hangup call you can use function TIMEOUT (absolute) or params of Dial command. buffers - W/O Change the channel's buffer policy (for the current call only) This option takes two arguments: Number of buffers, Buffer policy being one of: full immediate half; echocan_mode - W/O Change the configuration of the active echo canceller on the channel (if any), for the current. On the test, I set up the Asterisk CDR and the Asterisk CDR Viewer in Ubuntu Server. 167 countries available! Learn more. Understanding Inbound and Outbound Dial Peers. As such, the focus of development for this release was on core architectural changes and major new features. The number of active calls\channels in Asterisk. I use FreePBX - why do agents on a call show up with a different name? Completion of any call in the queue appears as Caller Completed. watch "asterisk -vvvvvrx 'core show. So you could filter just SIP/10 and SIP/11 for example. 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. Which should show you the following in the Asterisk console: Console verbose was OFF and is now 4. In this tutorial we will describe all commands available at the standard Asterisk version 1. I would have thought that at least the Digium phones could get the contact info imported with native FreePBX. cli reload permissions - Reload CLI permissions config. Data I would like to have: Num From , Num To , Duration, Codec, Context, Hold status ofc in realtime update I using node. 38 UDPTL support will be enabled, and T. the dialplan to request changes in the configuration of the active. Basically it shows all details of all active calls but I filter out only channel name and the duration of the calls but it there a way to keep active watch over it and if duration if over certain amount of time then execute a hangup request? I tried watch command but if I do that I cant use awk command? any help would be great. * Asterisk (default: 5038) and send an authentication request. In a simple scenario a single queue will map to a single public phone number. A couple useful commands to get you started are: sip show channels. Step 4: Hangup the active call. ambiorixg12: You need to use AMI, like in FOP. * By default, Asterisk system log in /var/log/asterisk/messages does not display called or caller numbers. Check the download page for the latest RasPBX image, which is based on Debian Buster ( Raspbian) and contains Asterisk 16 and FreePBX 15 pre-installed and ready-to-go. 8 my result was something like this i tried " core show channels verbose " this command to get output about active call. cli check permissions - Try a permissions config for a user. After that you will want to show the dialplan to verify. txt Copy to clipboard⇓ Download. Asterisk on the other hand is focused on being a PBX replacement and is architected to switch audio streams with all media going through a centralized system. To download an existing data file, go to Microsoft Teams admin center > Analytics & reports > Call Quality Dashboard > Upload now. AstriCon is the longest-running open source convention celebrating open source projects featuring Asterisk and FreePBX. All Asterisk users are encouraged to participate by leaving comments in the wiki to constantly improve the. active_calls,sudo -u zabbix sudo asterisk -rvvvvvx 'core show channels'| grep --text -i 'active call'| cut -d ' ' -f1 # number of seconds since last asterisk start UserParameter=asterisk. by dialplan using the Pickup or PickupChan applications. Install Python and pygooglevoice Based on the information given in [1], you can do the following to install pygooglevoice under the Linux prompt on the terminal connected to the Debian box:. Under balloons, set a checkmark on “Link Balloon text to specified table” and select the table you want to link to. console {device} — Generic console command. Try us out with a free call or see our services. Asterisk 13 Function_CHANNEL. Make a call to the assigned gvnumber, your SIP phone connected to Asterisk server should ring and can receive the call. The Asterisk Gateway Interface (AGI) Even he, to whom most things that most people would think were pretty smart were pretty dumb, thought it was pretty smart. Copy the channel name which you want to hangup. ” Each time Asterisk encounters a priority named n, it takes the number of the previous priority and adds 1. I have been reading the asterisk documentation for days now. These two channels will then be active in a bridged call. NOTE: If you are using Asterisk 1. all result show well but my problem is i want to show only call time after a call answered. 931 call setup message is received. feature show - Lists configured features. sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!. js --gulp to ingest the input file and log to the database. If for some reason thepeer is not registered and the IP of the peer is not known to the asterisk, above command will not work and CLI will not show any SIP messages. abort halt - Cancel a running halt. Step 3: Making a call to an arbitrary extension. The call is being made, but the endpoints never received the calls. To view a list of manager actions, navigate to PBX > Tools > Asterisk CLI and enter the command: manager show commands. After some time, Richard finishes his call and hangs up. Below is a list of the active calls for service being handled by the Volusia County Sheriff’s Office. restart when convenient: Restarts Asterisk when all calls are gone; show agi: Displays AGI commands; show applications: Shows all Asterisk apps; show application : Shows usage of a specific Asterik app; show channels: Shows all active channels; show channel : Shows information on a specific channel; sip debug: Enable SIP debugging. so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload (signalling, as well as ; PRI and SS7-related settings cannot be. at the end of the day, and how long they spend at their desks and on breaks. As you can see after the initial Invites the call is answered by the IP phone and a 200 OK message is sent to the PBX. All metrics are collected at once, thanks to Zabbix's bulk data collection. Only one action can be performed each time and the action packet contains the action name and parameters. csv file from scratch, see Upload tenant and building data. calls,UserParameter=sudo asterisk -rvvvvvx 'core show channels'|grep --text -i 'active call'|awk '{print $1}'. Richard is currently on a call, so Mark hears a busy signal. For see currently active calls you can use ForkCDR functions of CEL events (like some. By design, each active call gets mapped unto a channel. iax2 show callnumber usage -- Show current entries in IP call number limit table: iax2 show channels -- List active IAX channels: iax2 show firmware -- List available IAX firmware: iax2 show netstats -- List active IAX channel netstats: iax2 show peer -- Show details on specific IAX peer. console boost - Sets/displays mic boost in dB. Dial the two-digit extension number of the phone where you want the calls to ring. A new internal message bus, Stasis. It's a rough version and code is likely not optimal but hey, it works and maybe someone will find it useful. The template for monitoring Asterisk over HTTP that works without any external scripts. This is the online home of Asterisk: The Definitive Guide , a free book about Asterisk, an open source PBX platform that runs primarily on Linux. Cheap international calls from your mobile, landline or computer from 0. Note that the Asterisk command (in single quotes) is formatted for Asterisk 1. It is just a raw interface to Asterisk speed dial setup. You can safely use exit or quit commands to exit the asterisk CLI and leave asterisk running in the background. core stop now -- Shut down Asterisk immediately core stop when convenient -- Shut down Asterisk at empty call volume core waitfullybooted -- Wait for Asterisk to be fully booted dahdi create channels -- Create channels dahdi destroy channels -- Destroy channels dahdi restart -- Fully restart DAHDI channels. * Once this scipt is executed it will connect to the local port you have assigned to. Asterisk 12 is a standard release of the Asterisk project. As you may have guessed from the layout of this page, this book is published by O'Reilly Media. Here's the current output that shows aggregate count per second. You will have to write an AMI script that will connect to asterisk and subscribe for . To make a call, you have to perform the following steps, each of them explained in the next sections. Lastly, here's how the Calculated Columns #Active Calls, #Active Calls OUT and #Active Calls IN were determined: #Active Calls = CALCULATE (COUNTA ('CDR. You can change the data yourself to display the data. Standard phone: To forward your calls:. -Use the “ sh voice register pool 1" on CME to verify the registration process. A queue count is the current number of incoming calls waiting in a. If they do not reply on time, they will be considered unreachable. Step 2: Call Back The Number – Before drawing. In a nutshell, it consists of a list . I could not find anything simple for viewing active calls from Windows so I decided to make my own. Which will bring you into the Asterisk command-line client. This release is available for immediate download at. watch -n 1 \"sudo asterisk -vvvvvrx 'core show channels' | grep call\" - (Watch active calls on an Asterisk PBX Show active calls as the happen on an Asterisk server. There is no direct way of getting call time using "core show channels verbose" command. To connect to the asterisk CLI console. The channel is either on hold or a call waiting call. The Oracle® Enterprise Session Border Controller (E-SBC) uses the Digium Asterisk Hardware Device Interface (DAHDI) to support Time Division Multiplexing (TDM) operations. This is a place to read HTML version of the book (you can also buy a copy if you'd like to support the. SFS-270# show card Card Information. First of all i can recommend you use already created billings like a2billing. Examples Login - log a user into the manager interface. conf ^ ; ; ; DAHDI telephony ; ; Configuration file ; ; You need to restart Asterisk to re-configure the DAHDI channel ; CLI> reload chan_dahdi. In the My uploads list, click Download next to the file you want. 100:5060 N 12345 105 Registered Sun, 17 Nov 2013 07:47:21 1 SIP registrations. To upload the new file, see Add and update reporting labels. Type the following in the Asterisk CLI:-> pri show spans NOT Scenario: If a paricular span is not "Up and Active" (if it should be), then turn on span debugging in the Asterisk CLI to trace the D-channel message on that span in question. The Asterisk Development Team would like to announce the release of Asterisk 18. Current status is that it's not working but we can ping and traceroute successfully. By default, Asterisk uses ports 5060 for SIP and 10,000 through 20,000 for RTP, although that can be tuned with the rtp. Show active calls as the happen on an Asterisk server. This includes: A more flexible bridging core based on the Bridging API. by dialing the extension defined for pickupexten configured in features. Calls disconnected after around 30 seconds. Input file is watched rather than. I am setting up an asterisk call monitoring on zabbix 2. Add our script to file and do not forget to add empty string at the end of file: * * * * * /home/user/active-calls. system phone to forward calls manually: Press Feature 11. The feature is available for OpenStage 40/60/80. In the session details for each call or meeting, minor issues appear in yellow. When you change the dialplan in extensions. watch "asterisk -vvvvvrx 'core show channels' | grep channels" Watch number of active calls. Now dial extension 2000 with your phone. 8+) to verify the currently active application map. This document demonstrates basic techniques and commands to troubleshoot and debug VoIP networks. Monthly Subscriptions Sign up for one of our Subscriptions and get even cheaper calling rates to landlines and mobiles. Instructions on how to enable Asterisk Full Debug Logging Enable debug configuration in chan_dahdi. cli show permissions - Show CLI permissions. The range is from 0 to FFFFFFFF. sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE! 1. Note this will tell you the acceptable codes, and possibly the outgoing codec. watch "asterisk -vvvvvrx 'core show channels' | grep channels". [quote=“david55”]core show channel will give you the contents of the CDR answer time field. Last updated: Sunday, May 1, 2022 1:14 PM. Hi, Recently in the last few days my Asterisk system has completely gone to you-know-what. This page will automatically refresh every 60 seconds. show parkedcalls: Lists parked calls ; show queues: Show status of queues, see details here; show switches: Show alternative switches ; show translation: Display translation matrix ; soft hangup: Request a hangup on a given channel – in Asterisk 1. As soon as one of the requested channels answers, the originating channel will be answered, if it has not already been answered. The official source of documentation for the Asterisk project, this wiki is maintained by the development team that manages the Asterisk code base. You need to monitor call events to get information . If you need some your favourite text editor, do this: sudo EDITOR=vim crontab -e. 1 Centos 7 I currently use in linux commands to see active calls but I have to keep calling it manually Basically it shows all details of all active calls but I filter out only channel name and the duration of the calls but it there a way to keep active watch over it and if duration if over certain amount of time then execute a hangup request? I tried watch command but if I do. Would be nice to not have to have a SSH session open to look at active calls, should be a dashboard widget. Step 1: Initiate a Reverse Number Lookup – Conducting a reverse number lookup would yield results provided the number you’ve searched for is valid. Dial your two-digit extension number twice. This template was tested on: Asterisk. Once you are oriented to the approximate time, you will need to find clues that will help you to identify the call in question. If you want to debug the asterisk communication, stop the Asterisk service and start it using the following command. Right click on the view and press “Properties”. typing cmd $ asterisk -rx "features show". If qualify=yes or a numeric value, then asterisk will sometimes poke this peer by sending a "SIP OPTIONS" request to phones or other pbx's. Use the CLI command “show features” (“features show” in Asterisk 1. console answer - Answer an incoming console call. The syntax is: exten => s,n,Set (CHANNEL (echocan_mode)=off) The possible values are: on - normal mode (the echo canceller is actually reinitialized) off - disabled. This list represents some of the calls for police service being handled by officers of the Clearwater Police Department. My question is, how to blind transfer the phone call to B. Execute the netstat command alone to show a relatively simple list of all active TCP connections which, for each one, will show the local IP address (your computer), the foreign IP address (the other computer or network device), along with their respective port numbers, as well as the TCP state. A useful powershell script that I made to monitor active asterisk calls. Active IAX2 Channels - How many active IAX2 channels. I do not think it is a simple problem that can be fixed by parameters. Watch active calls on an Asterisk PBX. Our documentation and many Asterisk users speak about channels in terms of "calls". A call can be one or more channels. uptime,sudo -u zabbix sudo asterisk -rx "core show uptime seconds" | grep --text -i "System uptime:" | gawk '{print $3}'. Use the -n flag on the watch command to modify the refresh period (in seconds - default is 2 seconds). When i restore my server, i will need to sync it with asterisk (meaning… i will need to check with asterisk what channels are active as soon as it restarts). -> pri intense debug span X (replace X with the span) Note: Asterisk 1. In the Asterisk community, this feature is called "Busy Lamp Field"; sometimes the term 'Direct Station Selection' is used for the same functionality. This project site maintains a complete install of Asterisk and FreePBX for the famous Raspberry Pi. Configuration of Asterisk call parking; What ext. WaitExten (dialplan application). Only the channel information are displayed. This is indicated by the LEDs in an FPK. Our strategy is user and application scalability which is what sipXecs delivers inside and out. Via the command line of your server, issue the following commands: asterisk -r. Command to get the number of active calls\channels in Asterisk (there is . In line 8,1, modify 704 to reflect a voicemail box that is active on your system and that should be used for recording messages from unwanted callers. asterisk -vvvvvrx 'core show channels verbose'. number of actions for the client to use and each of them is decided by the module in Asterisk server. To check protection rules active in iptables: iptables –L –v. Creating a conference room is trivial, requiring only a few lines of Dialplan. The best way to do so is to utilize a Reverse Phone Number Search. The channel is set up based on SIP protocol. You can get more information about a manager command with: manager show command. Step 1: Initialize the Library. 0 resolves several issues reported by the. This command will show all the . At that prompt you will see live activity scroll by. When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the PJSIP channel driver will return busy as the device state instead of in use. it can be done, you need to monitor calls and grab the call duration and channel name and once you find the one who match the duration condition . This makes it easier to make changes to your dialplan, as you don’t have to keep renumbering all your steps. Hanging up active calls in Asterisk PBX. The calls accumulate in a named agent queue (or just queue) and there can be more such queues. Once 'sip show registry' showed up the trunk as registered however it didn't show up on web console as active registration. xxx for trying to get access to your server: iptables -I INPUT -s xxx. Anybody that can send me in the right direction?. The feature description uses always the same example background with fixed names, numbers and IP addresses. ; tested for call waiting and hold features Huawei E1550 has a next bug: ; When local side hangup any call including held call ALL other calls LOST sound ; When remove side hangup active call ALL other calls LOST sound ; Please double check true this for you or not ; If true reduce usage of this useful features exten => s,n,Dial(Dongle/g1. Calls with all relevant statistics are saved to MySQL database. asterisk -vvvvvrx 'sip show channels' -- does not show any data outout. Note: You can use the Asterisk cmd Goto () application to jump anywhere into the extensions conf , or Macro if you would like to be returned to the other party after macro execution, or use Asterisk cmd Transfer () for. You also can use AMI events+dialplan for track that (like fop/fop2 use), but tat is not easy way. To show current connect calls if they are SIP endpoints or trunks. CDR (Call Data Record), allows you to keep statistics on call activity in the MySQL database. On the Cisco SFS 7008, an asterisk (*) designates the active controller card from which you have initiated your CLI session. Not needed a ssh session, there should/could be Admin->Asterisk CLI in the GUI. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP/T. core show channels This command will show all the active channels in your server. conf I have set up the following command: Code: UserParameter=asterisk. This is usually only for SIP trunks because a phone registers to Asterisk, not Asterisk registering to the device. conf in the Channel section create a variable called: wat_debug= Below are the possible options to use for wat_debug: all. I will not go over option 1, but instead focus on option 2.