Freepbx show all extensions. but they can not influence each other.

• Freepbx show all extensions The nine primary extensions on our PBX are configured to light an LED for an extension when another user is on the phone (BLF. 1 and on the other I have 2. Main questions are: Does it take system resources (I don’t see double users though, only extensions)? Do I need WebRTC if I only use SIP (not planning to use browser)? How to disable those 99 The ‘convert2pjsip’ command is available in FreePBX 15 running the core module version 15. After looking on my network diagram I remember that I used DHCP filter for work VLAN and that Android device with extension and softphone used randomized MAC, upon set static MAC FreePBX stopped displaying those I just installled the FreePBX distro with the SNG7-FPBX-64bit-1904-2. 3 which is the older ser I have call pick while on the newer server 2. We’re using Yealink T48S phones. There is around 11 extensions My question now being, where in freepbx can I go to look at which extensions have enabled this feature, In Asterisk CLI the "database show cf" command will list the currently active call forwards. About 6 of the extensions will not register a contact with the AOR, so you are unable to place an inbound call to them. 06, with FreePBX 2. Also make sure that your sip passwords are at least 8 character long mixed case alpha-numeric nonsense. Applications / Modules. rpmed: Yealink Phones have these settings: Try this: account. One approach is to put a ‘dummy’ route after the emergency routes, which matches only the restricted extensions. I have added 3 extensions using the bulk extensions module to add them. There is a standard inside the HFA protocoll of the F650 which allows all pickup group members to see on their phone display, who is calling one of the other pickup group members without any sound. Same behaviour as listed below. Extensions remain with an ‘Idle’ state even though they are offline. conf should include an include line that leads to extensions_custom. Honestly, I think I’ve stumbled upon a bug. Is this possible? I figured out how to remove I have imported extensions via bulk handler. 4, three users, and three Cisco SPA303 phones. On that device (310) I also have another extension (410). I am a bit surprised that once a call is held, then the other extensions do not show the line in use. How can I accomplish this? I don’t see an option to set the route in the extension tab or add extensions to the route tab. All extensions are advertising with disconnected extensions: FreePBX-FSI*CLI> sip show peers Name/username Host Hey we are currently getting a lock up where no one can’t make or receive any calls. 10 lost connectivity with the VoIP service provider and with all the extensions. FreePBX and the phantom extension. Calling from outside to each extension is fine, calling between each extension are also working fine. I am able to see PJSIP contacts via CLI just fine so I know the AOR exist. All of our production systems have the exact same setup and they work beautifully; no problems whatsoever. the only problem i have now is that i I newly installed a FreePBX 2. Here is the original thread - Some extensions remain with idle state, BLF trouble Pretty much exactly the same problem. However this one only works when it wants to. At the moment all incomming calls and redirected to Misc Destination to a mobile phone. All your normal routes are placed after that. Want to group extensions into different groups like: Group A (dept A), Group B (dept B), Group C (dept C), Group D (dept D) etc Then want to create some rules to say: Group A can call all groups Group B can call only Group B Group C can call Group A and B but not D etc Is there any free or commercial module that Is there any way in the FreePBX Website for an administrative user to toggle or configure DoNotDisturb and CallForward settings for a I find it incredibly frustrating that the freepbx webUI’s “extensions” page will show an Well do not do that, whatever changes you make will be overwritten anyways. @GSnover I beleive I have done this before, but on the 2nd phone when you check voicemail it doesn’t accept the password from the first extension and when you put the password from the 2nd it then says you have no voicemail. org/display/FPG/Call+Forwarding+User+Guide. Hi Everyone, I’m running FreePBX 15. Thanks in advance! Hi there, we came up onto a problem as we are now moving from a Siemens F650 to the FreePBX Distro. These are extensions or trunks that have already been deleted. Create one or more extensions manually and export them with Bulk Handler, to see the format your . The other states seem to be working We have the server and all the wired extensions on the same dedicated subnet. Twice this week we have come in for work and all our phones are showing disconnected/not registered on the face of the phone, but “sip show peers” shows Asterisk thinking they Outbound call concurrency limits is an excellent feature, something i think we can all agree we have been waiting for a long time and its absolutely amazing. conf, directly or indirectly, Note that no files are read during a call; they are all read during a configuration reload. SIP extensions with new hardware also fail to register. What we are having is that certain extensions lose their registration and then re register. I have tried various settings in the field and then grep’d all files in the /etc/asterisk dir and do not find anything (this is after saving and applying the change). 95 and finding that setting the “Outbound CID” in the extensions general tab doesnt seem to be doing anything. conf to disable it but directmedia parameter is only accepted as individual endpoint parameter and I can’t rewrite Each time a new instance of your app is registered, just create an extensions. Need help. Yum update, Mondale update, system update, all show I’m up to date. Hi Folks, I’m just curios how big company FreePBX admins (as the rent out admin guys at Sangoma for sure do) handle 5000 extensions 35 locations 250 trunks or what ever? If I want to compare 2 extensions its an awfully work (clicking around in the GUI and waiting for the page to be loaded many times over) to do. Our FreePBX in in the cloud, so no phones (hardware nor soft phones) are local. If I configure an inbound route to route the SIP trunk to an extension, the call goes directly to Hello, Currently I am using Freepbx 15. I know that it is not a network problem because the other line that is registered on the same phone does not do that. Setting it in the trunk or the outbound route works as expected, but I Freepbx advanced Settings: Display CallerID on Calling Phone = No Display Dialed Number on Calling Phone = Yes. After I did I noticed that when I go to the extensions page they do not show and I get several “error” banners in red on the right hand side. Added SIP extensions (CHAN_SIP). Yes, I’ve searched and read several discussions on this topic. under trunks / extensions on the Hi, Hope all is well. Prior to upgrading from v15 to v16 of FreePBX, this option only showed their extension number. 22. With I have been tasked to remove all Caller ID from our Front Desk phones for inbound calls. 38. I only run 40 extensions, 22 trunks and 4 locations The complaint is that they now have to select their extension number from a list in the dropdown box. We are running chan_pjsip only. Test calls internally and externall (outbound) are also succesful. These calls do not show up in the CDR log. 62. 81 CO/PSTN Matrix Gateway -192. Current testing network topology is flat (all one VLAN). 11. 0. 1 GSM line issued from a USB dongle 1 SIP trunk issued from a provider. 13 on another box, same result). Got a site with a total of about 50 extensions. When I connect my inbound route to a ring group, and call my trunk, all my static agents (all Hi I have set up freepbx to call all extensions if the extension called does not answer incoming calls. If the customer dislikes the discontinuous ranges, we use 1000-1099. 6. I have a ring group and I would like that when one of the extensions receives an incoming call, all the others receive an alert of the call, allowing the service also by the others, referring to the visual BLF alert, could inform me of New purchase and setup of a ClearlyIP 716 appliance. Sadly, I don’t see it in my freshly installed v16. The quick background: Our organization has two departments with different DIDs on different trunks. How many extensions for a user? We have some users setup with multiple extensions: Office desk phone (hardware handset), home phone (hardware handset), office soft phone (PC) and mobile soft phone. 2. 17. Question 1: How do I see that history? Also, 2: In my User profile I have other extensions on other devices checked. There are more than a handful of “scoreboard” systems out there that can do all sorts of extension and queue status displays. 3. FOP2 manager version is 1. I’m really digging the idea of API behind freepbx upcoming versions. I’m looking for a way to see a list of all registered phones and what extension they’re registering to as well as their LAN and WAN IP. I’d like to see all the active calls in one display. After configuring freepbx and enabling Flash operator Panel, the Panel shows all extensions like there are online. 40 D Yes Yes A 5060 OK (20 ms) 2001 Looking for feedback to the best solution to do this. So, for example, if you configure a Mitel 6737i to connect to a single FreePBX extension, the phone will show that Line 1, Line 2, Line 3, and Line 4 are all registered to that extension. We only have two analog POTS line at this time. Do I need to form a “Pickup group”? I can’t find a pickup group feature with ASTERIX (freepbx) gui. Hello, I want to show all now i want multiple phones share the same extension, and all the phones are independently of each other. I have the CNAM updated by Sipstation and thats correct. If your particular user/extension number is 320 you can see all these settings by typing database show AMPUSER/320 at the CLI. Hi Using Sangoma S505 with EPM, (EPM > Extension Mapping) the IP Address says “Show AOR”, which does show the IP address, however how do i get the IP address to show up without needing to click Show AOR. Is there some kind of Is there a down and dirty way to just show all the extensions that are currently connected on my Asterisk server. PHP version 5. I’m just giving a taste of my office installation and Hello all, I noticed on some instances of FreePBX that under ‘Reports’ > ‘Asterisk Info’ > ‘Channels’ old data is displayed. I am new for FOP2. Is there anyway to do that in FreePBX? If so, can you point With chan_pjsip, the current preferred channel driver for VoIP, with SIP being the current preferred VoIP protocol, there can be multiple registrations for the same FreePBX “extension”. sh Hello, We are moving to a FreePBX system from our current * system, which is a base install configured exclusively through the config files. Both have 100@default, both show the light showing they have voicemail. I have the free pbx web admin up and running, have filled in some of the settings. conf so that instead of using many static inbound routes for our 1000+ DID’s, we’re doing a database lookup with MSSQL and sending the calls to specific Queues depending on the DID configuration in our call center app’s database. 4. 24 PBX Distro: 12. conf, but all I ever get is ‘State:Unavailable’ when doing core show hints I use 100 for parking as I have other dial functions using 7 as a prefix. I beleive that I saw that Chansip was being deprecated, so I converted all of the extensions to PJSip and upgraded to Freepbx 15. I got the (default) message that extension was Unavailable. 6 and Asterisk 1. There are many other attributes that control features such as FollowMe or VmX Locater™ and others. cid Hello, I want to show all the extensions that have an active call recording option I searched the database but did not find a table containing this figure. I am using FreePBX 14. 18. So that, they can easily dial and also, can see which client is Hello, Ok so I have a very small business with 20 extensions. Hi All, I’m trying to setup routing of my Gate intercom (which connects to FreePBX using SIP) to call BOTH extensions (which is me and my wife) when the intercom is pressed. After additional attempts we now have: WHAT WORKS: Internal extension to extension. We have a “poor man’s” admin setup where if someone calls an extension we have it ring multiple SIP phones. They are normally able to pick up the phone with in a certain amount of time. x. Now also, I looked at endpoint manager The first extension shows the IP address of the phone: Phone works well inbound and outbound. conf is 6. General SIP Settings: External Address=(my external IP Address), We only require an internal voip on our local area network, we do not have an Internet connection nor do we want to go outside of our LAN. Though, all imported extensions have callwaiting enabled. I am wanting to set two specific extenions (207 and 208) to use one particular Outbound Route and all other use the other route. FreePBX Community Forums Extensions with call recording option. 0 I was looking for central phonebook (CID) directory for all kind of extensions including hardware phones and soft phones. On the user’s phone it will list extensions like “100”, “101”, “1000”, “200”, etc On some, it showed the extension as “‘hi’ OR ‘x’=‘x’”. g. Although I cannot add voicemail details, every time I do add these settings it “forgets” them. Just Learning Hub / Tutorials / FreePBX / Add Extensions (V16) FAQs. I have all UCP settings set to yes in the user manager. (‘rasterisk -s Hey there. 25 All modules were upgraded. sometimes you have a fresh sever and you are looking forward to create like 100 or 200 or even 1000 extensions on the freepbx server in one operation . This is the same with SIP and IAX2 extensions. There are 4 remote phones Hello, I use Distro 14 with Asterisk 16. This topic was automatically closed 31 days Hi everyone. After restart freepbx virtual machine all extensions and trunk couldn’t register. 0 and FOP2 version 2+. Advance Thanks for Help Hi all, I am having the same issue as this post, which didn’t get many replies so rather continue an old thread I’ve opened up this new one. "Returned from dial-one with nothing to call and DIALSTATUS: CHANUNAVAIL" Originally, I setup a NAT on pfSense for UDP 5060,10000-20000 to forward to my FreePBX, but, my extensions wouldn't even register at that point. I am new to this. 65-13 Asterisk 13 and FreePBX 2. All the extensions seem to be screwed up. Following this guide: PBX API > RESTful I was able to create a quick php script to call pjsip extension status (based off of asterisk info contacts) <?php namespace FreePBX\\Api\\Rest; use FreePBX\\modules\\Api\\Rest\\Base; class Status Forgive the newbie question I have no background in telecoms and zero experience with VIOP systems but have been asked by my company to set something up for them to replace the antiquated phone system in the office. say that you want to create all of those extensions with specific Hey everyone, I have a production system running FreePBX 14 with Asterisk 15. The other areas of the Asterisk Info page are up to date (e. For instance Bob wants to call Anna at extension 111. I have a doctor’s office who is convinced he needs the old style call appearance/presence on all phones so you can see who is talking. I working on configuring and testing the FreePBX and extension configuration last night, after hours. I have many chan_sip clients cooperating flawlessly with Inbound and Outbound routes, ring groups, behind NAT etc including another IAX2 server and some SIP trunks in both of sides. So Bob call’s the main number September 10, 2018 between 7:50AM and 9:59AM EDT, my FreePBX system running version FreePBX 13. 2. I have 3 phones configured and intermittently I get something like this in the log files: [2013-12-20 13:14:09] NOTICE[1852] chan_sip. Tutorials: Add Extensions (V16) Step : Log in to FreePBX. I tested Most IP Phones will show multiple “lines” registered to the same extension on their web interfaces even though there’s actually only one registration. system (system) Closed May 15, 2020, 5:37pm 5. Thanks, Robert. Is it possible to only show a missed call on the extension that was originally called. Looking for a clean way to implement that with Polycom phones. FreePBX Community Forums Extensions registered on Asterisk. When i am login as “100” According to the call logs, everything writes like a normal call, as if it doesn’t even touch extensions_custom. : Registries). These extensions and the FreePBX are all on a routed WAN with each other with no NAT or firewalls in between Here is the asterisk console output when dialled from another extension(2101) which is exhibiting no issues otherwise. Is there a method Fresh install of Freepbx from iso on a ESXi stack. I push the voicemail button on the phone, and no joy. Now, I have a concurrent issue. This trunk is set to “InterOffice” which means the 3-digit Extension CID is preserved when calls are routed to the Central PBX. What I would like to achieve is to have an inbound call directly forwarded to an extension. The sip show peers shows the extension is active. agi: Starting New Dialparties. Functions as intended. I have callwaiting is disables by default. (23ms / ALL other extensions display correctly. most of the Aastra phones are pre-configured at the factory to support Asterisk right out of the bx you just need to assign extension, id, password and server. No luck. Enter your credentials. 1) running on ubuntu 6. So then I did another Hi, I am currently using AsteriskNow which include this FreePBX. Also, are there any variables that I can use in the dialplan to view all hangup handlers? (I know you can do so from cli core show Hello everyone, im new to asterisks! Downloaded and installed FreePBX distro 6. Installed FreePBX 13 with the main goal to create telephony station for my small dental office. There is a Nagios Module that can query the extension state and you can use that do display status. they just use the same extension number. Almost everything works fine, but from any extension pressing *65, it will all announce the biggest extension number. 28. Same issue. FreePBX Community Forums Call Forwarding missing from the UCP on just some extensions. Is there a simple way someone knows that we can either add in the users extension to a field (maybe the otherwise unused “userfield” getting tagged with the extension?) Ideally we’re looking for a way to do a simple search and pull out the data to Is there some kind of reporting or maybe a UCP panel that can show all extension information? FreePBX. I have 3 phones (1 Aastra 6757i, 1 Aastra 6731i, and 1 Polycome 335) and all phones show up in the EPM. Both extensions are created and registered: pbxtest*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 2000/2000 10. 1. I’m at a loss. Thanks How can i make all jitsi clients in the office to load automatically all extensions and their 15 the name of 14 is coming up. 8-2104-1. I initially opened this over on the Asterisk forum, and the findings there led me to open a bug, but I was asked to put the issue in these forums instead. Example ext 100 ext 200. A quick glance shows if the device is registered and from which IP, with the added bonus that you could click the IP address to immediately browse to the device GUI. These error, warning and notice start to show up in the I’m having problems with a few extensions These extension go directly to voicemail, and do not ring the telephone through a DID call, or from an IVR, or direct dial from another extension. dicko (dicko) May 27, 2020, 6:05pm 11. How do I go about removing an extension manually? Rudy. 11 and Asterisk 13. Command Options: fwconsole convert2pjsip [-a|–all] [-r|–range RANGE] To convert all chan_sip extensions to chan_pjsip: [root@freepbx ~]# fwconsole convert2pjsip -a Converted extension 6040 to PJSIP Converted extension 6041 to PJSIP I have a fairly simple setup with PIAF version 1. Kafluke (Kafluke) February 19, 2019, 2:01pm 1. The password in 100 is 1234 For my current understanding FreePBX uses in its „Extension“ mode one specific user per extension You have a single extension number with two devices, so everything (since it’s all tied to extensions) should be “right there”. I have one question, if one already has had a PBX server with many extensions is there a setting or an easy wait to set this Outbound call concurrency limits for all existing extensions? Hello there! I’m using RasPBX in Raspberry Pi 4. When a call comes in, it rings all of those extensions. I discovered that when I called from extension to extension. I found it somewhere (dont know where anymore) and adapted it to my needswith limited coding capabilitiesbut it seems to work so far. Thank you very much! We have a doctor’s office environment with one call group for all extensions. Confirmed behaviour on both server installs. 2 and about 40 extensions utilizing TLS for PJSIP and SRTP for call audio. I have a client who wants to somehow view the call history of all extensions for answered and unanswered calls. I am looking to build a Asterisks / FreePBX system to use as a prop phone system. From a shell , depending on your need for detail. for ext in $(rasterisk -x ‘sip show peers’|grep -E "[1]"|grep OK|cut -d "/" -f1);do rasterisk -x "sip sh hi, i need to migrate about 500 extensions and inbound routes from freepbx 2. Changing database entries is case sensitive, but listing them is not. FreePBX 16. 191. Also I tried to find a global parameter in pjsip. *45yyy*xxxx - Toggles the Agent in a SPECIFIC Queue and creates a hit for BLF for that specific queue to show them logged in. Sort by: I’m running FreePBX Ver 13. Carrier is Quantum Voice. FreePBX Version = 15. 6, FreePBX version 2. Is there a way to have the page sound on all phones without adding all extensions to a paging group? My concern with having to explicitly add an extension to the paging group is that we’ll forget this step when adding new extensions. I have freepbx version 12. 3 and o2. I have a production FreePBX 16 (Asterisk 20. Just in case, somebody finds it usefulit checks every 5mins (cronjob) /root/trunkcheck. Like I said, it has about 40 hello sir thank you so much for reply. If no one answers the call then it shows as a missed call on all extensions. c: Request ‘REGISTER’ Hi to all. 9. I created several extensions and installed 1 trunk succesfully. sentinelace (sentinelace) November 19, 2019, 5:44pm 1. Hello All, I used to use the following command to display all the phones, brand, extension and firmware version. 7. Hello, It looks like I’ve got a corrupted extension as when I click on the extension, a blank screen appears. When I click on History, I get prompted for my extension (310). The exploiters looking for victims that have their PBX open and use extension numbers for passwords, or something easy to figureout. Note: I have used the same extension numbers and same passwords for all the extensions so that I wouldn’t have to go one by one and re-register every extension with our New-PBX. Also, this extension doesn’t show up in asterisk console when performing “sip show peers”. I’ve been working on this all week and come up with a big nothing. For example, if your disallowed extensions are from 11000 through 19999, you might use: I’ve installed FreePBX on a fedora core machine. Running FreePBX 14. If I have multiple phones connected to one extensions (multiple AOR), I can’t arrived to send a this message to all the phones. Hi, One of our remote VOIP phones is suddenly getting multiple calls from extensions that do not exist. I have extension/extension groups 100 000 - 199 999 and 200 000 - 299 999. On the same PBX other extensions show the options. So you don’t > core show channels SIP/2514-001636c6 2514@from-internal:1 Ringing AppDial((Outgoing Line)) channel request hangup SIP/2514-001636c6 Requested Hangup on channel ‘SIP/2514-001636c6’ Hello Community I have this one extension that for some reason will just not display the Show AOR button in EPM. However, when I go to “pin” the ext to the phone they do not show up in the EPM. Or just use the call forward feature Or make a ring group and put the external DID ( I assume) in the group like 13334445555# and not that the # sign indicates an outside number to call. 1, Freepbx 2. 8. I created a Inbound route, with ANY for DID/CID to route the No. All extensions are part of the same template using the same designated model. I now have several extensions that show as online in Asterisk Info, and show as registered on the phone web Good afternoon. Phone one dials into a feature code that ORIGINATEs calls to each of the other extensions and then all end up in the same conference. However, I’ve tried multiple IAX2 and SIP clients and none of them FreePBX Distro 6. I have tried sendrpid and sendrpid here you got my config on trunk: host=192. The logs show: Connected to Asterisk 13. How I can make rule that only allow internal call between 100 000 - 199 999 extensions but not can’t make call 200 000 - 299 999? I have understand that I need to make I have a CSipSimple extension (2002) which will not ring when registered. iso. However, other phones in the company should receive caller ID, as normal. If someone calls my extension, they get a message that the call is I am really confused I have 2 asterisk servers both under module admin say they are up to date but on one I have a core of 2. No outgoing calls required. [split from this thread - mod] Hi Lorne, I see that the predial hook does not work when the target extension is offline. Just a display notification. they just have the same extension number. 65-31 I have changed the port setting in the extension module but have not been able to connect using that new port. What USED to happen was all of the extensions would show as ringing; in other words, the LEDs would flash indicating ringing on each phone. Moving two servers from Elastix to FreePBX 2. I’am using 1 Grandstream GXP 1630 and 3x Mizodroid on Android phones. FreePBX 14. On one of my PBX How to view active current call on any extension in FreePBX Connected to Asterisk 16. On the PJSIP Extension page, you'll define the extension number in the "User Extension" field. This normally happens around the lunch hour. These are still present on the Info page and are displayed as Offline. FreePBX. 44 Asterisk Version = 16. But if I call from extension to extension (7002 to 7000) the call goes directly to voicemail. Here’s the deal I’m trying to create a little demo applet that basically makes some phone BLF’s randomly show activity for a demonstration. sng7 Asterisk Version: 13. 9 and asterisk 1. Be sure to give this a friendly name in the "Display Name" field, and a password in the "Secret" field. Thanks. I would love to get a professional in to do this but being in the events industry, there is no money for that until Covid-19 goes away. 2 currently running on freepbx (pid = 2459) core show channels. agi: . Applications / Some how we got off the track. The phones are successfully logged into the system and can make outgoing calls, however incoming calls are not getting routed properly. What I’ve done failed in a way that I’ve not seen any coverage. What I need is for all other extensions (only two other extensions in this case) to ring when the Hi, Any help people can offer would be much appreciated. yair_pc (Yair Pc) June 28, 2020, 3:10pm 1. 0) to which 30 PJSIP extensions are connected on the same LAN. Incoming calls work as I want. sorvani (Jared Busch) February 20, 2020, 4:07pm 13. 16. 65-31 I have Good Afternoon, I have started the testing phase of FreePBX. In researching, it seems the DevState command is the way Hi guys, I found some old threads here that mentioned this issue. Hello I run FreePBX v14 on Raspberry and i set 2 trunks. 5 currently running on FreePBX-Gui this blocks all extensions from using any outbound routes unless specifically enabled in the Outbound Route’s Once you are on the Extensions configuration page, click "Add Extension" and select "Add New SIP [chan_pjsip] Extension. I did a full reboot of the system already but that didn’t help. FreePBX 15. 7 logs spam [2024-01-29 01:13:09] NOTICE[15782]: pjsip show auths return No objects found. FreePBX ARI Framework 2. This is an issue because we only have that one number, but multiple extensions. Still fairly new to PBX servers. When I look through the asterisk logs, I see the following: 37719 [2021-09-27 15:03:42] in various contexts via extensions-custom. If one extension puts a call on hold, then we would like to be able pickup from any extension. FreePBX Community Forums Extension Module Port Setting. Utilize FreePBX's web-based Click on the "+ Add Extension" drop-down menu, then choose the "+ Add New SIP [chan_pjsip] Extension" option. 5. normal phone icon on the phone; Thew second line, shows an icon with an X on the phone, outbound IS OK, inbound , nothing No IP Address showing in endpoint manager Hi All, I’m a student helper working part time in a consulting company. x type=friend trustrpid=yes sendrpid=yes&PIA qualify=yes nat=no I had a working FreePBX 14 server that I wanted to update to 15, then 16. Please where can be the problem? edit: I had to logon with all the phones and after logoff FOP shows the Dear all, I have a huge problem that I cannot seem to resolve, and I hope someone can at least guide me to find a starting point. conf. Voicemail indicator now shows I have waiting voicemails on the mailbox extensions. Given our currently low staffing, I have all queues set to ringall, and all extensions are in each agent list. I was trying to build something like Hello. The phone generally works just fine and it is on the same network as the PBX. 9 and making them HA with DRDB and Corosync. 3 (Also tried with 1. The Gate rings a single extension no issues but wont call the ring group I created a ring group with my 2 extensions (1003 and 1005). sng7 Asterisk Version: 17. Is this the recommended way to configure a FreePBX? So I understand the desire to get and show more details for the extensions but why get rid of the response time of the extensions in EPM? When I was using chan_sip you could bring up extensions in EPM and see every Hi, I have been fighting various days with this problem without finding the cause of the problem, I work in a company where they a have a FreePBX setup in Vultr, I have almost no knowledge with PBX server before, only a lot of experience with Linux, routers, switches, and many other network configuration related experience. 88 and higher. This is for both internal and external incoming calls to these phones. I have a new Asterisk install (v1. If a import contacts to freepbx will jitsi show them directly? as it does for the FreePBX when you make a call from 1 extension to another extension based on the extension name in FreePBX. 159 installed. My question now being, 1. for the system, it doesn’t have enough room in the Trunks section to show all 23 (technically 24, but who’s counting that D-channel, huh?) trunks using my PRI card. That is 4 separate extensions for this particular user. So the extensions_custom. I’m setting up a new FreePBX system with close to 90 extensions. csv file with one entry and execute fwconsole bi --type='extensions' extensions. rstebih (Rudy Stebih) February 18, 2011, 2:34am 2. freepbx. They’re registered and can make outbound calls, they just can’t receive calls. I’m at the stage of testing the environment, able to login to the admin console, get through the menus, read warnings, update modules, unban the IP through the console (happened during testing the virtual So, you should put your emergency routes first, so that any extension can reach emergency services. This worked for a little while, but now we have two cordless extensions that do not ring when the ring group is called, but do ring if you call the extension direct. 37 PBX Distro: 12. I want my client’s contact number shared with all of my 20 employees. When I execute ‘dahdi show channels’ I get Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service 1 from-pstn en default In Service 2 from-pstn en Hello, i have a fresh install of freepbx distro. This will include Direct Calls, Ring Group Calls, Queue Based Calls, IVR transferred calls, etc. Now I need to disable this option because I need the RTP streams going through the pbx, but I can’t find any parameter in Freepbx to do it. 24 and for my own reasons I changed PJSIP ports to 5080 and I have UDP & TCP enabled. 5 I am starting to see this with specific extensions at specific installs - Yealink phones (so far just T42G and T48G) are coming un-registered from Asterisk even though the phone thinks it’s still connected - under it’s status it shows registered, but Asterisk says no: Phone: Register Status - Registered Asterisk: 6065/6065 (Unspecified) D Yes Yes A 0 I have a central PBX which is connected to a remote office via a dedIcated VPN trunk. extensions. i n my organization i have free pbx installed on local ip 192. say that you want to create all of those extensions with specific i am looking for a module, that shows graphically the status of all extensions or a selection of extensions (registered/available/ringing/in use). 38 - all modules are up to date. So I built a fresh system (updated it) and did a restore of pbx settings. I have created one user “100”,Group “Test” and allocated all extensiosns under that group from FOP2 admin portal. 48 (Current Asterisk Version: 13. I added an extension to my PBX, for some reason it works for one minute after I sync the settings and then it just rings as busy anytime you call into the extension. Now I am looking solution how to restrict calls. Hello, I’m newbie to FreePBX. 10 Asterisk 1. admin - config edit - The system can only create one extension per incoming route/DID (000 1234567). 1 i don’ have it working and cannot seem to get it to work. After 15sek or so I want to add additional extensions to Hi everyone, I am a totally new user, looking for a guidance. Seems to have issues with IVR transferring to Extensions. My system details: Rasperry PI 3+ OS: raspbx NO trunks at all. 1 version. If I try to create all extensions and manually enter the DID it says “route or extension already in use”. I'm using the latest version of FreePBX and Endpoint Manager. Display Name: Enter I have a weird incorrect Caller ID Number issue I cant figure out whats causing it. All my extensions are PJSIP extensions. All works fine all incomming calls are properly The CDR reports out the box seems a little unwieldy when you start trying to use it to analyze end users phone activities. We are moving over to a FreePBX system first of the year and I’m in the process of getting things setup and ready for the move. 240 with 4 FXO and 4 FXS ports BSNL WINGS SIP service (India) Pfsense+ firewall with NAT and firewall rules Dynamic public IP allocation by ISP - attached to DDNS domain by pfsense. I have successfully configured my external SIP trunk, and I can call out using that trunk. Is there any predial hook that does capture these calls? I tried grepping ‘predial’ but it did not return anything. The telephones themselve work perfectly otherwise they can call other extensions, dial outside lines, etc Any ideas would be appreciated if there is any additional I built a new FreePBX server using the latest ISO download, updated all modules, and applied all system updates. Depending on the show we could add extensions for different gags. The Asterisk/freePBX will not register, shows no sign of trying in the logs and is just doing nothing. So I Second, I set FreePBX->Applications->Extensions->8005->Advanced->Mailbox to “8003@device”. dialparties. These extensions are Fanvil X4G phones, so there is no NAT or strange configurations; they are all in the same FreePBX Yes, exactly. Also, two trunk lines, one inbound route that rings all extensions, and one outbound route. Asterisk 16. The call history on the phone shows incoming Hello My setup FreePBX - 192. so instead of trying to overcome issues i had with that before i am going to approach this entirely different. User Extension: Type in the user's extension number. But real three ext from them are offline. In the Panel that comes with FreePBX which show all of the extensions, trunks, etc. Optional regular expression pattern is used to filter the peer list. Today I can send SIP SIMPLE IM message between extensions but only to one AOR contact of the PJSIP extension. Where would I find this information in FreePBX 13? FreePBX Community Forums Get registered phone info. 24. 6. I am asking about the UCP, mostly History. PBX Version: 15. I Greetings, I recently updated my dev system. But I have one phone that will not show their BLF button. Is there some way to clean this little area up to show the right stuff or am I stuck with it showing only some of Hi We have the FreePBX 13. My message_context is correctly set for I’m configuring FreePBX for use across two sites but I cannot get one extension at one site to call another extension at ”. Now it shows all extensions on our deployment, including virtual extensions being used as group voicemail boxes. I have 11 extensions on PBXact: 15. Testing with X-lite softphones and the they are unable to register with the server. I’ve seen this question before, but wondering if there’s a better workaround since all the other posts are from a decade or more ago. Normally I am able to get it back going by using “core restart now”. Everything went smoothly. 2) with all updated (modules and system), i have a number of pjsip extensions that are showing as offline on the Asterisk Info area, but these are also extensions that have been totally deleted from extensions area and user management area. 0 Any help would be great. I’m having this issue on all of the extensions in the company today so I need to solve this fast. Is FreePBX provides any REST API to do this or need to purchase any other tool. Is this a c In the good ol’ days, with chan_sip extensions, the FreePBX admin could browse to EPM Extension mapping and see a list of configured devices which also showed the device IP address. 11, Zaptel 1. You can get them all using “pjsip show Module of FreePBX (Extension Settings) :: Creates a list of all extensions and their current settings for CW, CF, CFB, CFU, VMXB and VMXU The feature is enabled on an extension by dialing *72 as per https://wiki. Basically I want the phone to show all 3 SIP extensions with their current status on the screen like the image below. I have agent timeout set to 10 seconds to hopefully rotate them In all cases create an confusing outbound rules for international calls such as long prefixes and/or pin codes. The cordless unit we are using is the Yealink W65H. Any help would be awesome. 3 7 of the extensions are all at the corporate office and their caller ID is the main company number. (See attached image). 65 with aster11, configured it, [/code] Having trouble with BLF hints and sip presence for a couple of extensions. We do have settings enabled: EPM > Global Settings > Extension Mapping IP Addresses = ON EPM > Global Settings > Extension Mapping Phone With both the above for "What isn't working" asterisk full log shows this when any of the extensions call each other. However, if the biggest extension got a voice mail, all extensions will have that voice mail tone We have a ring group on 602 that is set to ring all phones in the building. We have 40+ 9133i phones. Hi all! I’m doing a test calling from one extension (2000) to another (3000) in FreePBX 13. Depending on driver being used, you could return the status of all or just one extension with: Asterisk -rx"sip show peers" Asterisk -rx"sip show peer 1000" Asterisk -rx"PJSIP show aors" Asterisk -rx"PJSIP show aor 1000" You could then slice up the output to get what you need then email yourself or whatever. agi dialparties. I added the Endpoint Management Module. The voip phones on the cell phones (via wifi on the same subnet) unregister All is good. How do I see those histories? On all but one phone every BLF shows up, even the phone's own extension. How to disable callwaiting for all extensions at once? I use FreePBX 13. when I try to call this remote SIP extension from other working extensions, asterisk tells me ‘the ext # is unavailable’, also in FPBX Panel the remote SIP extension while not completely greyed out (as the other SIP extesion is which is not registered at all, for now I try to make working just one of them) it doesn’t look as brightly lit as the rest of extensions, also on *45 - Toggles Agent in/out of ALL their Queues *45xxx - Toggles Agent in a SPECIFIC Queue *45*yyy - Toggles the Agent in/out of ALL their Queues and creates hints for BLF to show they are logged in to what Queues. Is there a down and For 20 extensions or fewer, they are numbered 100-119. All of the extensions were Chan SIP. I’m swamped right now and don’t have time for experiments. FreePBX 2. However when I add an extension in the FreePBX web gui it shows up all fine. I have a Trixbox and FreePBX set-up in our office with 9 phones and a 10 external phone numbers (the 10th being a catch-all group for the main number). I have changed extension to be shared among 4 different phones - Cisco Hello, By default pjsip extensions are configured with directmedia=yes. This allows the extension CID to be displayed on all internal calls. We’re special needs school and need “Page All” capabilities. I have created 18 exensions in freepbx. I Easiest is to use follow me and put that other # in and remove the ext. csv and then fwconsole reload. Using softphone program right now X-Lite, which when we try and call either of the two extensions set up gives back the message: "number is We’re using FreePBX/Asterisk in a call center environment and have setup the extensions_custom. The grandstream settings panel shows the phone is registered with the PBX. 168. I looked in our FreePBX settings and see that under Settings – Asterisk Hello community, I’m looking for a way to add extensions to a ring group after a delayed time without displaying a missed call message on the already ringing extensions. FreePBX Version is 4. We’ve done this by appending a letter to the base extension, so that calling 7771 dials SIP/7771&SIP/7771a, etc. regards, gunther One of the features of the Panasonic system is that a line on hold will show up all extension/phones. 194. JessicaRabbit January 1, 2016, 8:33pm 1. 10. We’re using When you dial an extension internally it goes to the from-internal context and looks first for an extension on the PBX, if it does not find one it’ll try to match a dial pattern in the outbound routes. 120 . Share Add a Comment. The latter I would like to Monitor extension like 40001, 40002, 40003 and 40004 on the same time for example, how the commands will be? usually i will use sip show peers XXXX , to monitor 1 extension. I figured I just hosed something up on the system as I routinely try things with it. i have read that this is not possible with a backup from within freepbx for some strange reason. 20 there are about 50 extensions configured and i need to setup a remote sip extension so that user can make a call to extension number using internet not using any vpn is this possible if then please guide me This is a script, which monitors SIP peers (trunks & extensions) and sends an email, if a peer is unavailable. If a phone was Hello @Stewart1,. This is a S705 phone with latest firmware and this is a PBXAct with all latest updates. The dial pattern for the GSM Dongle is set as follow : no prepend ; prefix = 0 ; match pattern = . While I have a bit of knowledge on general computer systems, I am completely new to Freepbx and Asterisk. Fraser. 12. gregorywest (gregory West) June 2, 2018, 8:48pm 1. . When I calls on my FreePBX phone I see incoming number on my softfone, but I need hiding wthis number an type: unknown or everyone else for all extensoins in my FreePBX How can I do this? Please help. Extensions and reinvite setting for all extensions set to No. just like: i have a extension number 2001 in two or more company（A and B and ）. Try running “sip show peer 1000” at the asterisk CLI followed by “dialplan show 1001@from-internal” and provide the output for both. For up to 100 extensions, one option is: 100-119, 290-299, 390-399, , 990-999. But when you dial an extension from a IVR it goes to the from-did-direct context which I don’t think will try to go to the outbound route. The solution was to disable WebRTC in the user management. Under “pjsip show endpoints” dont show the Yes all transfer modes are supported on the Aastra 480i, it’s even covered in the user manual on page 19. 3CX extensions register immediately (including existing working freePBX extensions which then reregister with freePBX just fine when pointed back at freePBX) 7 Hello, I have what may be a dumb question. Any extension can pick up and resume the on hold call. When a try in asterisk cli sip show peers, there is everything ok. c: Peer ‘563’ is now Reachable. Using PJSIP. For example: We have two extensions which should ring first. softphone - GSwave Lite on android and on iOS I want to know whether i can create a custom These fields provide the vitals to manage the given FreePBX user/extension and associate it with any number of devices. Log file from unsuccessful registration is this: [2016-10-26 08:40:10] NOTICE[32445] res_pjsip/pjsip_distributor. ) That works fine. Looking at the debug, these lines are showing as busy, but I don’t know why. 5 and asterisk 1. However, a call placed in the remote office to a public (Outbound) number, will also be sometimes you have a fresh sever and you are looking forward to create like 100 or 200 or even 1000 extensions on the freepbx server in one operation . when i try some command it return, sip show peers [like pattern] Lists all known SIP peers. Still, for several of the queues, it’s just ringing one person at a time. 5-1807-1. 40. I have installed FOP2 in my centos and its GUI working fine. System is Asterisk 1. csv file should have. I need to hide incoming CallerID’s for all my extensions. I need to create Extension from my application and get list of extensions also. but they can not influence each other. The existing extensions I created before updating the framework/core show FreePBX Community Forums No Device Options when adding extensions after recent update. Access the FreePBX command line interface (CLI) using "pjsip show endpoints" to retrieve a list of all extensions along with their associated IP addresses[1]. Third, I try using “*97” in FreePBX->Applications->Extensions->8005->Advanced->VoicemailExtension. 4 to freepbx 2. And, Indeed all previously registered extensions with the Old-PBX did register with the New-PBX, no changes were needed. idb jvegon glb neqvf vsgeu exojor tcoxd rkctyw yhuvj gvgo