Freepbx pjsip multiple endpoints. I currently use Devices/Users mode.

Freepbx pjsip multiple endpoints I tried to set the pstn answer delay to 10, but it didn’t help. 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. It provides a resource for assigning multiple trunks via SRV addresses, and more options. Manually doing so obviously works but the PBX doesn't seem to like that idea. I’m using Freepbx 16. Can post the result of pjsip show endpoint YOURENDPOINT that shows where the contacts are set to 1 and there is more than one contact registered? Vengeful_Blade (Vengeful_Blade) February 26, 2022, 12:57pm 3 Hello all, Is it possible to register the same line multiple times? Specifically this is being done on Yealinks. All ports are properly forwarded through my firewall to my PBX (presumably). To allow push notifications to the mobile, a Bria server registers as PJSIP account and wakes up the mobile when a call comes in, then the app takes over. I have no trunk set up so i don’t care if someone hacks into it. Then you will be able to add the affitional registrations as lines 2 and 3. PJSIP also provides three main components of real-time multimedia application, i. Jun 2, 2022 · In Asterisk SIP Settings, chan_pjsip tab, under transport 0. Then failed to authenticate. I copied the addresses from a notepad and saved those, but when I Jan 10, 2022 · Hi, I am using a PJSIP trunk with Gradwell UK and they have a list of IP addresses where I should allow traffic from. I am trying to establish SIP trunk to third party SIP PBX . Mar 4, 2023 · I have recently built a new FreePBX server running FreePBX 16. Also I tried to find a global parameter in pjsip. That’s because FreePBX, the world’s most popular open source IP PBX, gives users the tools to build a phone system tailored to their needs. Be sure to give this a friendly name in the " Display Name " field, and a password in the " Secret " field. Starting 2 weeks ago, extensions will randomly go unreachable, then again reachable: [2024-02-13 2… This page will outline how to setup remote phone BLF's using PJSIP between two PBX's which will monitor the device state of remote phones. May 10, 2022 · Continuing the discussion from Third party webrtc phone with FreePBX: Does anyone (@lgaetz ?) have a custom dialplan example which I can implement in the extensions_custom. For some unknown reason, I see multiple registrations from the Bria server and (related or not) mobile won’t ring or it will take a few seconds before it does The Endpoint Table of Contents The Endpoint Instantiating the endpoint Creating the library Initializing and configuring the library Creating one or more transports Starting the library Shutting down the library The pj::Endpoint class is a singleton class, and application MUST create this class instance before it can do anything else, and similarly, once this class is destroyed, application Apr 20, 2015 · Now, with the understanding that chan_pjsip can have multiple endpoints for a single extension, which would simplify my dialplans, ring groups, queues, etc. Thanks. Once the Status for each city displays as Avail, you can begin making test calls between the servers using a phone connected to each PBX. 201 Below is my configuration Below is the messages received on CLI: freepbxCLI> freepbxCLI> freepbxCLI> [2023-10-17 08:11:31] ERROR[22407]: res_pjsip. I am developing a SIP endpoint and I am using MicroSIP to test it. I also have a desktop phone registered to the same extension. But when I go to associate the extension to a new phone that extension does not appear on the drop Detailed manual for FreePBX PJSIP installation Aug 2, 2022 · Is your SIP-channel-driver in settings/advanced setting on “both” or non pjsip only? Can you find the new extension at admin / config edit → xxxsip. I’ve enabled this feature for now to allow my PBX to receive calls. conf file to also ring a WebRTC extension together with a primary extensions… without using the follow-me/ring group function? Thanks a lot for your help. Other way is use 3 asterisk with chan_sip each one binded to own ip. Is this the recommended way to configure a FreePBX? Seems complicated and also a management Feb 16, 2022 · C’est à dire que l’appel sonne en même temps sur les deux postes (bureau et Dect) david55 (david55) February 16, 2022, 12:41pm 2 chan_pjsip should handle this easily, see, for example, TIPS - Multiple endpoints on 1 pjsip extension | The VoIP-info Forum Mar 9, 2019 · BTW, given that Twilio sends calls from multiple IP addresses (therefore requiring multiple chan_sip trunks), why didn’t you just use a pjsip trunk for Twilio, where you can simply list the addresses in the match field? Feb 16, 2018 · I have come to Fusionpbx (Freeswitch) from the FreePBX world. This made me want to enable PJSIP and move forward with connecting multiple endpoints to one extension. e. There is also nothing that restricts you from having an endpoint, an aor, and the extension used to call the endpoint share the same name. Reports- Asterisk Info-Peers does show IP address and RTT This occurs on 2 different servers, all up to the latest build. Chan_PJSIP doesn’t have the concept of “allow guests” which means it won’t just default to a peer if nothing else matches. TIA! Feb 28, 2024 · I am trying to set up an OBI device as a PJSIP trunk. It has been working very well. That is 4 separate extensions for this particular user. 34 and we’re getting 488 Not Acceptable here with SRTP specifically on PJSIP. I’ve tried installing new certs, migrating to a more powerful server Jan 18, 2022 · 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”. conf to disable it but directmedia parameter is only accepted as individual endpoint parameter and I can’t rewrite them because May 26, 2022 · Chan_PJSIP doesn’t allow anonymous users to auth unless they either are match by an IP or there is an anonymous endpoint that has been created. After an hour of anger, I figured out, that BLFs start working again, when you set a presence state Nov 14, 2020 · I am trying to upgrade from a very old asterisk now installation running happily with a couple of Cisco 7970 phones in SIP mode to FreePBX (15. I dont know where to whitelist 5060 Nov 16, 2019 · In trying to figure this out I stumbled across an interesting configuration issue. I have found out where I can set multiple different proxy set ID’s so I can have one use the default port and the other use the IP of the freepbx server with port 5160, but I can’t figure out how to set the different ports to use specific proxy set ID’s Feb 22, 2023 · Good Evening, We are having a bit of a stage issue, We have just setup a new FreePBX system and we are having a bit of an issue, (Multiple Phones) - We have a couple of Old mitel phones and we register them using End Point Manager - They have registered successfully and work for internal/external calls. Sep 13, 2019 · Hello, I’ve received a report from a customer that when they park a call on their primary device it rings back to their secondary device once the timeout is reached. After it completes, tried to run: *CLI> sip show peers No such command ‘sip show peers’ (type ‘core show help sip show’ for other possible commands) *CLI> module show like sip Module Description Use Count Status Support Level 0 modules loaded *CLI> pjsip show endpoints No such command ‘pjsip The document provides a step-by-step guide on setting up a SIP trunk in FreePBX using CommPeak SIP account credentials, including accessing FreePBX, configuring trunk settings, applying changes, setting up outbound routes, and testing the configuration, with troubleshooting tips using Asterisk log files. Dec 18, 2015 · Looks like endpoint manager-extension mapping is not reporting IP address and status on endpoints that are connected via PJSIP, CHANSIP endpoints are showing this info. Switching off SRTP or making it optional works fine, TLS still works at the signalling level but SRTP is a no-go on PJSIP. Mar 12, 2020 · I can see a codec list for the trunk setup under pjsip settings. My trunks and Oct 2, 2020 · This isn’t actually PJSIP behavior itself, it’s Dial behavior - the same as if Asterisk had been told to dial two chan_sip devices. This was very helpful at times. signaling, media features, and NAT traversal, among other things that have been taken care of by PJSIP. FreePBX is an open source user interface (UI) for Asterisk, an open source telephony server. Sep 13, 2023 · Is it possible to have multiple (2 or more) PJSIP phones on the same extension all using the FreePBX built in OpenVPN server? Initial tests produce an IP Conflict as the VPN Server hands out the same IP to both test phones. However, I believe you will need some manual edits to the pjsip configuration. Feb 14, 2024 · Have about 400 extensions behind a SonicWall; we have no control over the SonicWall. If you don't want to do it manually look at using the endpoint manager module or the open source version that is free. We’ve Sep 15, 2017 · When you have 3 endpoints registered on the same extension account, then they all ring when you dial that extension number. Possibly, sending 717xxxxxxx instead of +1717xxxxxxx will help. I am havin gissues with my freepbx server. Is there something particular to setting up one extension on two different endpoints in the FreePBX gui that I’m missing to cause this ‘only most recently booted rings’ thing? Dec 12, 2020 · The PJSIP module doesn’t currently present that information, but it’s certainly something we’ve heard that people want. All of the calls are working correctly (inbound and outbound), but I cannot make the caller id working. 100 trunks, this is not practical and you need to explore other solutions. In this world we used Pjsip extensions to have multiple registrations on one extension. Would anyone happen May 7, 2019 · Hello, By default pjsip extensions are configured with directmedia=yes. I’ve successfully registered three devices using the same extension. Any help on this? Mar 20, 2024 · Hello, I can’t say I have much experience with FreePBX, but it seems I have a rather odd problem. May 23, 2024 · Part of the problem here is that for chan_pjsip the extension configuration defaults to having Direct Media enabled in FreePBX. Any good guide? I found multiple ones for SIP and tried but they are not working. Initially, both phones were ringing correctly, but now, only one rings at a time. This is very annoying when the desk phone computer softphone, and mobile Jan 9, 2020 · Hello, I have been using openvpn to connect my Bria softphone on my Samsung mobile and Laptop to my pjsip extension. I only need to disable access to the current users MicroSIP and Jun 17, 2016 · In summary, it seems that NAT detection and NAT features (chan_pjsip’s “rtp_symmetric”, “rewrite_contact” etc. Modify the dialplan on the phones themselves to prepend a “site code” to 911, then catch that code in an outbound route that delivers the proper 911 caller id. SIP Trunking SIP Trunk Interconnection Guide FreePBX FreePBX PJSIP Configuration Configure FreePBX Sip Trunking with PJSIP IP based Configuration You have the option to configure it using IP-based or SIP-based SIP trunking. FreePBX Configuration: SipSetting module (v14. 7. Jun 28, 2018 · Using a Chan SIP extension, you can easily set those port numbers in the extension configuration. My SIP trunks (to providers) are connected via dedicated NAT 1:1 IP as non-NAT trunks via special low-latency but narrow-band line. 12. PJSIP is set as 5160 in FreePBX. Asterisk Background Publishing Extension States Exchanging Device State PJSIP Environment Oct 8, 2019 · Hello! I have a Linksys SPA3102 as a pstn trunk added to freepbx with a pjsip trunk. If I enable pstn cid for voip cid on 3102, I’m not get the inbound caller number at freepbx, I get only an annonymous. Jan 28, 2024 · pjsip show aor 7005 pjsip show endpoint 7005 pjsip show contacts and post the output (as quoted text, not as screenshots) Is extension 7005 registered? If so, what happens when you call *43? When you call an external number? If anything appears in the Asterisk log, paste the relevant section at pastebin. Created a PJSIP Extension, on SIP Settings enabled only TCP Transport and NAT information, I’m using Zoiper as a client. In that world you can dial extension 101 and it will ring all devices that have 101 registered In the Freeswitch world it will only ring the first device registered for ext 101. I have a PJSIP extension, which I am connecting to the SIP endpoint I am developing. Jul 15, 2020 · FreePBXEndpoints asterisk, configuration, pjsip, extensions jwmb224 (jwmb224) July 15, 2020, 2:59am 1 I have not been able to find any recent topics on this anywhere on the internet, so i’m not sure of the appropriate terminology, so i’ll just describe the use case i’m going for and hopefully you guys can weigh in and point me in the right direction. 24. The users have an s500 phone, a cell phone using Groundwire, and a computer using MicroSIP client to connect. But in PJSIP, I cannot find any way to set the port number on a per-extension basis. This is a very requested feature, but not something native to FreePBX. Here’s the situation: When we first add an extension to a phone, everything works fine: the phone can make and receive calls without any problems. Asterisk still showing this error: WARNING[29555]: res_pjsip_registrar. Anyone encountered and/or solved this problem before? Take care, Nate Dec 1, 2020 · the PJSIP max_contacts value is extremely useful, as it allows multiple endpoints to register to the same extension, and allows for “Sustainable” VPN use to connect to the phone system – as the endpoint IP may change if the VPN connection is closed/reconnected. 1 using chan_pjsip, this use to work find on Asterisk 13 using chan_sip, but I am unable to get it fully working on chan_pjsip. TRANSPORT (provided by module: res_pjsip) Configure how res_pjsip will operate at the transport layer. c Oct 2, 2015 · I am planning on moving over from freepbx 12 to freepbx 13 this weekend. I’ve two extensions registered as PJSIP, when they call each other, there is no audio. com and post the link here. 240 - connected to FreePBX using Peer-to-Peer trunk Pfsense+ firewall with NAT and firewall rules Dynamic public IP allocation by ISP - attached to DDNS domain by pfsense. When everything is working fine, they appear like this in the PJSIP Peers list: Endpoint: 112/112 Not in use 0 of inf InAuth: 112-auth/112 Aor Mar 20, 2020 · We use this for hot-desking and multiple contacts with pjsip so phones have the same extension. Aug 5, 2024 · Chan_sip was using port 5060 I also brought the old freepbx server back up and changed the chan_pjsip on it to 5060 and converted the extensions over to pjsip and the FXO ports on the SPA8800 were able to register successfully on the old server. c:1068 find_registrar_aor: AOR '' not found for endpoint '200' (<ip address of other PBX>) So the blank AOR is causing the problem, but how to I assign Oct 31, 2019 · The Continuous mode, in particular, is used when a SIP server supports multiple registrations at the same time. softphone - GSwave Lite on android and on iOS I have been trying to switch matrix gateway from chan to pjsip… i am able to make outgoing calls on the pjsip but Feb 7, 2018 · A basic concept with chan_pjsip/res_pjsip is the endpoint. After I had changed to chan_pjsip, the BLFs stayed dark. admin2all (admin2all) August 3, 2022, 12:44pm 43 Nov 22, 2021 · We have converted a FPBX box from chan_SIP over to PJSIP (for the endpoints) … and now is there a way from the Asterisk CLI (to figure out) what type of phone the PJSIP endpoint is (Yealink, Polycom, Fanvil, etc)? In the past: we could have used the command: sip show peer (and) we could see not only the model of phone, but its firmware: very easy to do. If you need e. The extension is configured as a PJSIP extension and does work on the first phone I associate it with. Fairly simple and it works. When a new SIP request comes in, res_pjsip needs to identify which endpoint the request is for. You can do pjsip show contacts which will show you all the contacts and their URI’s, RTT times, etc. 0 (ignore the logs, we’ve since upgraded) on FreePBX 14. I now have a need to remove devices from an extension due to an employee termination. Mar 9, 2022 · Previously on an older version of freePBX using chan_SIP endpoints we would use the option allowsubscribe=no to not allow subscriptions however I now see that pjsip has allow_subscribe=yes set by default. I had to turn on core set debug 9 to see that it it’s even trying to look at the INVITE because there’s absolutely no other events I got the versions from the upgrade tool. 27. This prevents me from receiving the calls unless I allow “Allow anonymous inbound SIP calls” and “Allow SIP Guests”. edit: Oh, you want to round robin it? Mar 14, 2022 · It is chan_pjsip that allows a different Asterisk side port to be used for different extensions. xxx. Occasionally my openvpn will cause problems and my Bria softphone will become unavailable and fails on registrations due to the max contacts limit. Please advise. 2. If one rejects, the other continues to ring. I’ll try to describe the situation in detail. endpoint parameters), seemed to be ignored for the Internet-based endpoints until I listed the FreePBX server’s own local network as a ‘local_net’ (despite no phones actually being present in that ‘local’ network). 0 on VMware (Cloud Telephony), i have over 20 virtuals machines working very well. FYI - Switching to Chan_SIP with the same codecs, and it works perfectly with SRTP. I’m not seeing any outbound or inbound call traffic when I test, which makes me think I have a registration issue. 0 (udp), Port to Listen On should be 5060. 65) using pure PJSIP. See full list on crosstalksolutions. When i attempt to register it gets denied and stated no matching endpoint found. Feb 26, 2022 · That doesn’t sound right, we use that feature all the time on that same version of Asterisk (not the same version of FreePBX). conf that is generated by FreePBX, and for extensions there are no entries whatsoever. Some of the extensions ring multiple endpoints as configured in their extension while other will only ring one endpoint. The only port you can set per extension for chan_sip is that of the remote user, but, I presume that the remote user is registering, in which case the port number is taken from the registration, along with the address. Sep 1, 2021 · I have setup GoIP gateway which using chan_sip. So my questions, if a person is on a call Jan 12, 2022 · Late to the party here, but we want to move to PJSIP for our trunks. 76). For the security, reliability and available band Sep 2, 2024 · We’ve upgraded to FreePBX 17 and our configuration has worked except where we have two endpoints registered on an extension, the busy lamps don’t light. Can anyone else verify? Jul 14, 2022 · 2 particular units did register and were shown as Endpoints and functined fine initially… but then just disappeared from PJSIP Show Contacts / Show Endpoints !! I can access their GUIs both from a Browser and from the phones GUI and try to re-register them but they will not register and I get “Failed to Authenticate” errors on them. You can get them all using “pjsip show endpoints”. Has anyone found Mar 1, 2022 · I’ve problem using only TCP protocol for Extensions. Apr 28, 2017 · Now, with the understanding that chan_pjsip can have multiple endpoints for a single extension, which would simplify my dialplans, ring groups, queues, etc. sng7 with Asterisk 13. Assuming you are using pjsip. Jun 2, 2015 · I’ve just setup a new FreePBX server (12. You don’t create an anonymous endpoint Sep 20, 2015 · Hello! I was testing the chan_pjsip driver this weekend, because I like the idea of sharing one extension on multiple endpoints. 156:5060 is now Unreachable. Nov 1, 2024 · My setup FreePBX - 192. RTT: 0. But every PJSIP trunk has an entry such as this: [trunk_name] type=identify endpoint=trunk_name match=sip_server_from_pjsip_settings Sep 24, 2025 · For some reason all of mt endpoints are unavailable so my calls are not getting routed to the routed to the correct context My trunk is named “2292341730” Trunk is set to qualify every 15 seconds and is successfully registered. endpoint 1 would have to park the call and then endpoint 2 or 3 could pickup from that parking lot. I’ve tried multiple variations of this page - One with the Authenticate ID being the same as the SIP User ID and currently trying it without it. 6+) FreePBX GUI has an option to configure Jul 29, 2021 · FreePBX 15 asterisk 16 PJSIP Sangoma D series phones I know there’s a way to set this but can’t remember if I’ve done it right (and I’m not onsite). For some reason one of our servers has started randomly dropping 1 or 2 lines at a time out of about 130 total lines. For 3 months everything was fine. The setup includes creating a trunk, configuring SIP or PJSIP settings, and defining credentials provided by the service provider. If you’re using chan_sip you would have to use the unsupported devices and user mode. I get no audio. Dec 2, 2023 · You can use PJSIP to have an extension on multiple endpoints, but this will not be a true shared line appearance. I want to register extension, say 1001, as Line 1, 2, and 3. Below is the registration information: Username: 23011936@siscomeg. They worked fine. 8-2208. For the Network settings, I haven’t tried much other than changing The “Free” in FreePBX stands for Freedom. For most users this is not convenient - it’s better to stop all others to ring Oct 21, 2025 · I think you need to define an explicit ring group, rather than the implicit one that happens with chan_pjsip multiple registrations. For example, it supports configuration options for protocols such as TCP, UDP or WebSockets and encryption methods like TLS/SSL. This order configuration is useful in PJSIP scenario where we have PJSIP extensions and trunks are coming from the same IP. The deskphone subscribes, but it doesn’t get hints. When I was using chan_sip, I had created some BLFs for other extensions on my Yealink T38G. Mar 1, 2017 · res_pjsip. I would try to create a new pjsip endpoint, which may be either listening in port 5160 or on port 5060 is sip settings are “both”. sng7 I am trying to connect my Cisco IAD2432 Version 15. We do this because we have extensions that appear on multiple devices. I’ve tried both the UCP password and the secret. Trunks act as communication channels, allowing calls to be routed in and out of the system. In the VPN Server I can create multiple certificates with various IPs. Dec 18, 2024 · Once you are on the Extensions configuration page, click "Add Extension" and select "Add New SIP [chan_pjsip] Extension On the PJSIP Extension page, you'll define the extension number in the " User Extension " field. Feb 27, 2020 · Hi I’ve FreePBX 15 with Asterisk 16. 81 Matrix Gateway -192. Communication start, the client connect correctly with TCP on port 5060 but UDP traffic start to be sent and there are no audio and the call terminate after 30 second for . 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. We have been unsuccessful thus far. Internal through my PJSIP extensions. How to I haven't worked too much with pjsip yet, so you'll need to google/test yourself, but pjsip allows you to have multiple devices connect to a single extension so dial (sip/extension) would ring all the devices connected to the extension. Once the call is answered at endpoint 1, endpoints 2 or 3 will not have visibility into it. 168. If necessary, define a separate route for each DID to direct calls to the correct destination (e. You can do it with chan_pjsip, by using more than one transport. It appears that other users are affected by a bug (feature?) in the conference bridge setup. But sometimes, on client side, the phone loose is registration, it seems we cannot register the phone. 13. I copied the addresses from a notepad and saved those, but when I Dec 15, 2023 · When an incoming call comes my trunks come as ANONYMOUS. Mar 27, 2025 · This guide will walk you through setting up your SIP trunk using PJSIP, a modern and versatile SIP channel driver in FreePBX. I’m just giving a taste of my office installation and what I am trying to say is that In this article we will explain how to configure a FreePBX PJSIP V13 Credentials Trunk with Telnyx. May 16, 2023 · Once all of your PJsip trunks are activated, you can verify functionality in the Asterisk CLI with this command: pjsip show aors. However, I am unable to get my Cisco register on Apr 26, 2022 · I want to use an extension on multiple devices at the same time and it should be active. I want every phone in the ring group to get the alert that there’s a vm in that mailbox and be able to access it from the vm app Dec 17, 2020 · Hey does any know how to configure SIP forking to soft-clients w/diff IPs (same subnet) sharing the same extension? I thought PJSIP supported this by default but was wrong. Mar 24, 2024 · I’m not sure PJSIP will work, so I have both enabled on my system (PJSIP and SIP) and have configured the trunk as a SIP trunk. 111. Can post the result of pjsip show endpoint YOURENDPOINT that shows where the contacts are set to 1 and there is more than one contact registered? Vengeful_Blade (Vengeful_Blade) February 26, 2022, 12:57pm 3 Feb 28, 2024 · I am trying to set up an OBI device as a PJSIP trunk. I am attempting to use this as a local service acting as phones for an office like setting. Jan 3, 2018 · I have a location that historically has always been one phone one extension. There is a dialing sound, but there is no audio trading at all Sep 4, 2024 · Issue with inbound calls using freepbx 17 pjsipe FreePBX Endpoints siptrunk, pjsip, freepbx, configuration fraddygil (Fraddy Gil) September 4, 2024, 8:40am 1 Nov 12, 2025 · Topics tagged pjsipnext page →Topics tagged pjsip An aor may be shared by multiple endpoints, or an endpoint may have multiple associated aors. You can setup multiple transport sections and other sections (such as endpoints) could each use the same transport, or a unique one. Our current infrastructure is that FreePBX is trunked to our Avaya Session Manager via ChanSIP with the following configuration: type=peer qualify=yes host=SessionManagerIPAddress context=from-pstn-e164-us keepalive=30 Can anyone help with how to translate the above peer trunk to a PJSIP Nov 24, 2019 · Hi, this is strange. 22 I’ve had a tech onsite to confirm this, and can’t see anything in the config that would impact this. endpoint. immensely. 11+ Or v15. Jun 21, 2023 · Hi, I’ve been trying to get a Grandstream WP825 to register on FreePBX. I read that allowing anonymous inbound sip calls is a security risk, hence I would like to fix this issue as soon as possible. That would mean any of your internal calls would have both endpoints trying to use Direct Media with each other and yes that can be impacted by the Internet connection they are using and other networking factors involved. Apr 11, 2022 · You can register multiple clients to a single PJSIP extension provided the ‘max contacts’ is set to a number equal to or greater than the maximum number of registrations. Apr 6, 2020 · Hi guys i have a freePBX 15 installed and i setup PJPSIP trunk on it adding the ip address on a remote PBX also, specify the fixed public ip address of that remote PBX and also set context=from-internal -Now this remote PBX will call the local extensions of my freepBX directly by SIP and teh calls works perfectly good audio and connected, problem it drops after 33 seconds tried multiple times Mar 24, 2020 · These are PJSIP extensions and the console shows the correct public IPs for the remote extensions. I have verified that the extensions are Aug 29, 2020 · Hi, We’re running Asterisk 16. If they call out side via trunk it works well. To really match the SIP RFC concepts, I think the section name given in PJSIP dialstrings should be that of an AOR, but I think it is actually always an endpoint. Ran asterisk-version-switch on FreePBX 14. 6. Used the hostname and the IP for it, the SIP Server. 1. (We were using chan_sip before so this is new configuration. I want to use pjsip instead. The system is rapidly growing and expanding, so creating chan_SIP extensions for each of what will soon be hundreds of devices spread across the state and then programming and maintaining Follow Me settings for each one wouldn’t be practical. All you need to do is use the PJSIP_DIAL_CONTACTS dialplan function to create the Dial application’s dial string to call each device. No outbound calls. However you should note that: Sep 19, 2025 · Topics tagged endpointnext page →Topics tagged endpoint Mar 21, 2015 · Not really. Apr 23, 2024 · I am trying to figure out with an AudioCodes MP-124 how I can register some ports with SIP on port 5060 and others on PJSIP port 5160. Nov 8, 2024 · Because the documentation is not clear to me, I’m not completely clear how the degeneracy is broken when you have multiple AORs for an endpoint, or multiple endpoints for an AOR. so has configuration option i. 33 System Version 12. freepbx Aug 25, 2023 · Hello, I am very new to the freepbx platform. One of the factors driving this is the ability to connect to an extension using multiple devices (e. This is on 12. 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. Aug 30, 2022 · Hi, Am I correct that I cannot have multiple endpoints with the same extension and haven them connected through the FreePbx VPN. So you don’t want anonymous users trying to auth to make calls, no problem. Aug 21, 2018 · I have a Sangoma UC100 that is using PJSIP and allows up to 3 devices to connect to each extension. Endpoint: 2292341730 Unavailable 0 of inf OutAuth Mar 24, 2023 · Hi guys, I need a little help, i’m sure it’s something stupid but i cannot find what it is. Jun 8, 2020 · We use FreePBX in Device and User mode. That list does not include any video codecs at all. And i use only yealink deskphone and Gigaset dect. a phone and a softphone), which apparently requires pjsip rather than “standard” SIP. I gave up on further searching and I know that this may be asked often but I am in a dead end. g. 0. com Jun 18, 2020 · Simplest way is use PJSIP channel driver with different endpoint section (each one to different ip). There’s an incoming trunk, incoming catch all routes, etc. So a single endpoint would list all the contacts for it. I upgraded the system to Asterisk 13, set up a test extension using chan_pjsip, and it seems to behave exactly the … Thanks, David May 1, 2020 · We want to not have to split the setup process like this in cases that require the same extension ringing in multiple offices/buildings and so want to just use the gui. Please select one of the options below to access the interconnection guide based on your preference. Jan 16, 2020 · Hello, I have what may be a dumb question. Table of Contents: May 12, 2023 · Trying to set up this incoming SIP trunk (pjsip) from audiocodes to freepbx. Mar 29, 2020 · I just curious if there is a workaround to have separate external IP specifically for remote extensions? The problem is that standard Asterisk setup assumes single external IP setup only that is used in SIP headers. Not sure how to go about Aug 5, 2022 · I am having a trouble for days to figure out how to configure pjsip trunk. , From: "NAME IS HERE" <sip: [email protected]> Next, the original caller ID standard required that the device be capable of displaying a 15 character name and a 10 digit number. The list is long, a total of 23 IP addresses Gradwell Customer Community and I am adding those through the FreePBX GUI, but it seems that there is no more space to add all of those under the Match (Permit) field. Now with PJSIP, it’s possible to have Apr 1, 2021 · PJSIP supports the use of contacts which allows you to register that extension on multiple devices. There is a file in the /etc/asterisk directory called pjsip_identify. "endpoint_identifier_order" to determine how res_pjsip will match the incoming SIP request against present endpoints. When I reboot one phone, the available status switches to Apr 12, 2021 · What would be the solution to run ad-hoc phones (multiple users for each phone) with PJSIP and extensions mode ? Jun 14, 2023 · hello And I have a problem with a single extension, the 300, it makes a normal call, records the audio, but does not receive a call. But the issue is that there is no audio after the call is answered. I can see no codec list for extensions though (I thought this is what the general pjsip settings are for?) May 13, 2025 · Check with pjsip logger that the selected header is being sent with the correct data, e. I upgraded the system to Asterisk 13, set up a test extension using chan_pjsip, and it seems to behave exactly the way I’m expecting. when I run the command: asterisk*CLI> pjsip show endpoints Endpoint: 300/300 Not in… Oct 17, 2023 · Good Morning Colleagues I hope you are doing good . I currently use Devices/Users mode. I’m not sure Jan 15, 2025 · FreePBX trunk configuration involves connecting your PBX system, VoIP provider, or another PBX. FreePBX is behind a NAT. Asterisk does not reply whatsoever to the INVITE. See *80 Intercom - No Audio from user who dialed Endpoints Jan 10, 2022 · Hi, I am using a PJSIP trunk with Gradwell UK and they have a list of IP addresses where I should allow traffic from. Compared to the older Chan_SIP driver, PJSIP offers better performance, enhanced NAT handling, and improved support for multiple registrations. That’s what having multiple registered devices does when dialing - the Dial application dials each of them. I set up a phone in Germany, it is a Mitel 6863, It registered like no ones business. 16. 6-1904-1. five trunks, you can set up multiple transports for pjsip. Also, “Find me follow me” works fine to different extensions, but forks both calls to the soft client with IP address A, while ignoring the soft-client with IP address B tied to the same Extension. Am I just not seeing it, or is there really no way to do this? And if not, how do you keep the extensions separate? Yes You need to increase the max contacts to more than 1 to let you have multiple registrations. Nov 21, 2022 · Hello, Currently I am using Freepbx 15. , if a FreePBX user dials 202, route it to extension 202 on MikoPBX): At this point, you should be able to log into the Asterisk console via SSH (use the command asterisk -rvvv to get to the Asterisk console). However, there are a couple May 11, 2020 · Here are the settings I have set: *I set the extension using the PJSIP port 5160. There is no registration or authentication, I have tried various both to add it via the FreePBX GUI and also adding Nov 27, 2023 · I’m experiencing an issue with two Grandstream WP822 cordless phones that I’ve set up on a single extension (PJSIP). PJSIP is an Open Source and separate extension of the Asterisk, and Asterisk derived systems. In this article we will go through how you can set up a SIP-trunk in FreePBX in a matter of minutes. Like others, (Reject call should stop ringing other endpoints (PJSIP)) I immediately observed that “rejecting” a call does not send the call to Voicemail, and the other endpoints keep ringing until the timeout is reached. Aug 31, 2022 · Hi, we’ve got several different freepbx servers, they were all updated to the same level, all modules are on the same version, and they are all managed identically with identical settings across the board. However, in EndPoint Manager, if I select an endpoint to use the VPN it only gives one choice for all all endpoints with the same extension. Judging by the Wireshark traces, the registration is successful (server returns 200 OK), but after invoking pjsip show endpoints from the CLI Nov 29, 2017 · Upon reading that chan_pjsip supports multiple AOR’s such that several devices can act as one endpoint you may think that’s a neat feature. I made sure that the extension’s password on my phone matches the secret in Freepbx’s extension. We suggest using PJSIP Apr 21, 2022 · Hello, I have small raspbx server running on port 5060 but I have 2 external endpoints/extensions that for security reasons I have setup on my router external port 5065 but internally 5060, so for freepbx/asterisk it is all under 5060 but for the endpoint I had to add xxx. So from what I understand, we can now change to pjsip and switch back to extension mode and a single extensions can register from multiple devices/endpoints at the same time. Once in the Asterisk console, you can run 'pjsip show endpoints' and you should see the new Crosstalk SIP trunk in an 'Avail' status (Available). Please can somebody let me know what is the pjsip equivalent of: May 6, 2021 · For two trunks with different source ports, set up one on pjsip and the other on chan_sip. Mar 20, 2018 · We chose PJSIP because of the need for multiple endpoints to be registered to one extension. I’ve got a virtual extension with vm to dump all “support” calls to if nobody answers the ring group. I recently added PJSIP devices, mostly softphone apps. FreePBX also has my trunk as an endpoint set to unavailable and no incoming INVITE requests get routed to it. All modules updated fully. Let’s take a look at how to configure FreePBX with VoIP Innovations using PJSIP trunks. ) Our Snom D715 deskphones subscribe and show BLF hints, but not for those of us with DECT handsets as well. Jun 5, 2025 · FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. May 19, 2016 · We have all our extension set to PJSIP in our FreePBX 13 with Asterisk 13. ca Password: mysecret Third Party SIP PBX IP Address: 192. pjsip show contacts displays multiple unavailable contacts for my Jun 13, 2024 · Connecting endpoints to FreePBX and experiencing problems? This category is for you! Nov 11, 2020 · Do you still have multiple registrations for the problematic extensions? My system does not use a conference bridge on intercom calls to pjsip extensions (with single registration). Sometimes it just goes directly to Voice Mail… As I was talking to that ext (when it decided to work), I got this (while still on the phone): -- Contact 2022/sip:2022@192. Remember this is PJSIP which can have multiple contacts per endpoint unlike Chan_SIP which is one. I’m sure it isn’t difficult with FreePBX, but that confuses extension numbers with endpoint IDs, so you may have to cope with having to define more than one extension as the source of calls from the user. This mode avoids any gap in SIP registration because the Bria Push server is always registered on behalf of the Bria Mobile client. The inbound number Sep 25, 2020 · 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. Dec 18, 2019 · I have a freepbx installation back home with port 5060 forwarded so i can register phones outside of my network. *I set the to use both Chan_PJSIP and Chan_SIP in Advance Settings. In SIP it was the last registered device but with PJSIP, in the extensions we’re having an issue with, it seems to ring whichever device it wants to ring. On the chan_sip tab, Bind Port should be something different. conf. If I run “pjsip show endpoints”, I’ll see all three contacts listed as “Available”. 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. But when you reject a call on one of them, then the other endpoints keep ringing. Don’t use chan_sip, which at this point shouldn’t be used anyway. 24 and for my own reasons I changed PJSIP ports to 5080 and I have UDP & TCP enabled. I’m trying to register the same extension on more than one phone. That’s ok, and when you answer the call via one of them, then the others stop ringing- that’s ok as well. Apr 24, 2025 · FreePBXEndpoints configuration, freepbx, pjsip Armand April 24, 2025, 3:00pm 1 Hello everyone, I’m currently facing an issue with Polycom SoundPoint IP 335 and Polycom SoundPoint IP 650 phones on our FreePBX system. 000 msec -- Contact 2022/sip Feb 21, 2023 · Basically, I have several endpoints that allow two Contacts because they have two physical phones in different locations (say, one in the office on their desk and one at home) – same extension number. but when the user picks the phone up and dials ext 2008 - It will not display the persons Jul 17, 2024 · Hi I’m using Bria Enterprise softphone app on Android and a couple of Ios too. If you still have trouble, at the Asterisk command prompt type pjsip set logger on make a failing test call, paste the relevant section of the Asterisk log at pastebin. Now some mobile users are going to be moving from one location to another. Nov 15, 2024 · Hello! I’m encountering an issue while trying to register my Polycom VVX 350 phone to FreePBX using PJSIP. I checked Asterisk Info and noticed that only one phone is listed as available, despite both phones web gui saying they are registered. Options? Asterisk Apr 30, 2016 · I’m running FreePBX 13/Asterisk 13 and have the commercial endpoint manager installed/licensed. 12 to go to Asterisk 16. Even when i restart the phone / dect Aug 7, 2020 · Stewart1 (Stewart) August 7, 2020, 6:32pm 2 Assuming pjsip extensions, try setting (for both) Rewrite Contact: Yes Force rport: Yes Direct Media: No and confirm that anything NAT related is turned off on both devices. xxx:5065 to access the router but when I do this the call drop at 31 seconds. This is easy with Asterisk. FreePBX is licensed under the GNU General Public Lic Feb 15, 2021 · I have 2 phones registered with PJSIP (ext 2022) The 2 phones are not playing well together… It’s 1 working the other not, or both not working. xzo aev wmatv esurpbyo cja maerp kxoigfr yoven modzt kkvb hehdy smdd hiwin peawmz ypiawm