Freepbx internal calls not working. Hello, I am new to Asterisk and FreePBX.

Freepbx internal calls not working The VPN server has a iptables rule that masquerades all outgoing traffic. 3 which is the older ser I have call pick while on the newer server 2. I also made the outbound route but probably something goes wrong with the route and the calls don’t go: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 > [INSERT INTO cel I can call from cisco to asterisk extentions and from asterisk I can call cucm extensions. Any ideas? Thanks . The softphone does not have any SIP provider registered and is used only for testing, but can be also used to receive calls. Internal calls are working but I can’t seem to figure out why I can’t make outbound calls on the same Flowroute trunk. The receiving phone rings, user picks up and then the call is immediately disconnected. I saw They are working fine with everything except when they dial an internal extension. Line 2 can only receive calls; Line 2 is configured with an italian provider. The I’ve just set up SIPSTATION on my freepbx system and I am able to send and receive calls. Only changes done to the system was System Updates. All calls internal or external go straight to the busy voicemail greeting. 40. Of course, redact your phone number and account I am having issues with call reliability with a MicroSip Softphone on a PC. 29. etengftw (John Ramborat) March 20, 2023, 3:38pm 1. I initially thought it may be a codec issue, but looking at a packet capture all phones are using g. I was configuring FreePBX and SIP Trunk from NTC Nepal. Inbound calls are succeeding. asterisk, freepbx. configuration. The PBX has a public IP address and is one of many within the same data center and is the only system having an Currently using: Asterisk 1. It does not appear to be a registration issue because the phone dials out just fine. FreePBX Community Forums Follow me suddenly not working on local calls. Is it possible to use this for internal calls only? We still do not have SIP or other connections yet Hi I have installed FreePBX 15. Nothing has changed configuration wise. 7 . 0/24 to a 10. No playback, no error, etc. They want to forward all their calls to their cell phones and did so using the Call Forward All feature on their Cisco 7960 IP phones running SIP firmware. Internally calls work fine, and i am stuck. Hi mates, Covid-19 has changed our place of work, so we are telecommuting from our homes. I quickly saw that it is not a good way, and it is better Hello, Since running the conversion tool to migrate from 32-bit to the latest 64-bit distro, I notice that external calls aren’t coming through. The external calls comes from a SIP trunk. I’m running FPBX-14. If they Calling out stopped working while everyone was out to lunch it seems. You are not the only one having problems. Support & Resources. I can not Hi I am using an FXS converter with an analog phone plugged in. Hi I got a FreePbx 2. That’s working fine, it’s simply bridging 2 trunks, one to Lync, one to Cisco. We need to know your FreePBX some setups such as Outbound Route and Asterisk Logs will give to us a bit more details. 18 PBX Distro: 12. When we internal dial, it rings perfectly fine, but as soon as you answer the call, it immediately hangs up. 11. 0) i made a trunk and an incoming route for incoming calls and everithing is ok. They can also receive a call internally from the same phones that disconnect just fine. I am able to Hey guys, I’m having a weird issue. Clicked the Apply button and made a few test calls from the said extension. 14. Ext 225 transferred the call to 221. 5 machine running shorewall. 🙂 I am struggling with making outbound calls to gamma SIP. My current situation is no outbound calls, get 480. I have set “CIDLookup” to “Internal” and have told both trunks to use this Starting yesterday afternoon, I have 2 sites that have grand stream phones connected to cloud based Free PBX and outbound or “internal” extension calls are unable to be madeI have attached a trace showing me making an inbound call and answering. 7. However, other phones in the company should receive caller ID, as normal. I’ve setup a FXS extension and set Record Incoming and Record Outgoing to “Always”. I checked with Vitelity to make sure there wasn’t a problem on their end and I was having similar problem with my freepbx that I installed using the scripts to convert stock cent-os 6 into freepbx distro. I have looked at other posts and have applied many of the remedies suggested. conf and discovered that the a line saying: The remote extension was working for a bit yesterday, and today I have no audio on the remote extension. It would usually land the caller on a phone menu, which I can see attempted in the logs but on the device making the My internal calls are working but external external calls not working and mentions ‘All circuits are busy now’, in asterisk cli prompt I can see res_pjsip_header_funcs. What I want it to do is return to from-internal and continue processing without skipping anything that should execute, but an include apparently isn’t designed that way. Is anyone able to help? I can see in inbound call in the full log, but not work out what is causing it. Hello, I am new to Asterisk and FreePBX. 0 Any help would be great. Everything was working fine until the systems stops processing calls, no internal calls and no external calls were Possible, I had Hangup Cause 41 and 16 on the Logs, someone has any idea why this At the Asterisk command prompt (not a shell prompt), type pjsip set logger on make a failing call attempt to e. If a call comes from extarnal, it is not forwarded to an external number (cell phone). I did a packet capture in the System Admin and have it in Wireshark, but I’m a novice and don’t know what I’m looking for. Here’s the Hello all. - Freebpx : Centos 5, asterisk 1. Endpoints. f to external numbers has to be enabled by a different way? (e. Zulu Client: v3. When a user goes into the UCP and selects “Call History” and then clicks on the play button the button turns into a spinning circle and nothing happens after that. 42) Created a pjsip trunk using credentials from MXOne Extension called “MXOne200” Then I created 2 extensions in freepbx (88888 and 77777) for testing. I can do outgoing calls without any problem! Quality of calls are very good. The usual tone you get (ringback) when the other phone is ringing isn’t there, but you can hear whatever is on the other end of the line when the call is answered. Outbound calls work fine. Providers. I am running Freepbx 2. What have I missed here? Context ‘from-internal-noxfer-additional’ tries to include nonexistent context ‘from-internal-noxfer-additional-custom’ all the 4 phones are registred with FreePBX and works fine (except from that one that I thought I did explain what I had done with the NAT. If the call is not reaching your PBX, verify trunking connectivity. localhost*CLI> module show Hi, we have a FreePBX Distro 4. I have been able to get the intercom working. On an internal call (extension to extension) FollowMe is not following, but is working on external calls. I have the problem on different systems. I setup a VPS with freepbx last August and I kept an eye on it to make sure that everything was working fine before going live in my company. 21. I’m planning to use the FreePBX just for home and internal calls Perhaps I am not making myself clear. The endpoints are Grandstream phones (1615/1760/2160), which are able to register with the PBX server successfully. They gave us a pair of IP’s and basically said good luck. The Call Monitor feature is simply not working. 0%, SL2:0. Calls from an outside line and calling outbound works, but within extension calls have no audio. Now the call comes in as Unknown Unknown (before it was coming in as John Doe 1-234-567-8910). 2, sames config (trunk and SIP) ==> same problem - Old Freebpx Box : same config (trunk and SIP ) but it works Ok I can make Outgoing calls but I cannot receive any incoming calls? This is probably because either DNS is not resolving the NTP server correctly, or the NTP server is not responding. Can you provide us a little help via Remote Desktop? Stewart1 (Stewart) May 13, 2022, 11:45am 4. This file has the correct channel: Channel: SIP/SBC2/+123456789 None of the calls get the 2024 prefix and I’m not sure why? I can see in the CDRs that SBC2 is used, but the configuration is Hello, I’ve created a custom click-to-call app for our internal web-based CRM to FreePBX, using Originate through AMI via a perl script on the PBX machine. I then copied everything from the context [From-internal-additional] located in extensions_additional. I understand how the MoH settings are tied to the routes (inbound/outbound). org and post the Hi all, I’ve just installed freepbx 2. conf i see: Hopefully this is something simple! I’ve just migrated to FreePBX 16. Then I looked for the recordings on /var/spool/asterisk/monitor. If I: Place a call Put that call on hold Call another number Click the merge button and select the call that was on hold Nothing happens. We got back from a company lunch and no outside calling. The incoming calls are registerd from asterisk, i think only the transport to the SIP phone is not working, but i Hi all, this morning our inbound calls have stopped working. I checked with Vitelity to make sure there wasn’t a problem on their end and they are not even seeing those calls we are trying to do reach their system. I have the logs pasted below: Hello, I have everything working except for outbound calls. In the network traces for failed calls, I see my end reaching out to the RTP server, but no return traffic. I want it to be strictly an emergency phone for 911 dialing. Normally I am able to get it back going by using “core restart now”. I can make outgoing calls but incoming call are not working. FreePBX software is free, however that does not mean that you have some right to free help. Upon Hi all, If anybody has seen my previous posts, I’m a noob and new to networking and PBX in general. 0/24, along with a new Fortigate firewall. VoIPTek (VoIPTek) September 22, Hello, I recently updated from Asterisk 1. 10 x86_64 server. In practice, i am able to call from remote (client over VPN) to local There is no sound on incoming calls only (the caller can hear us but we can’t hear them) however outbound calls work just fine and inter-office extension calling works just fine; also dialpad inputs do not work on the IVR as well. 6. My VoIP phone is Grandstream GXP1630. If the call is reaching your PBX you should see it. The Yealink T58 can dial externally and calls works just fine, no disconnect. I added my flowroute pop's IPs by CIDR and that allowed incoming calls to work. Also, not sure if this matters but I changed the default root password. This is accomplished because ext-local should come after ext-findmefollow in the include list of from-internal-additional generated in extensions_additional. i am using the polycom 560 phones and i am on the current version of freepbx. 34) to restrict some extensions from making outbound and internal calls. g. IP is 172. I added a new extension about 2 weeks ago - and that ext works fine since then all new extension - are not working properly we are able to dial out - no problem receive incoming calls - no problem but - access voice mail - is a problem We plugged in a new phone at this client. I dialled *60 (Speaking clock) and I can hear the time announcement clearly, Hey we are currently getting a lock up where no one can’t make or receive any calls. org and post the last 8 hex characters of the link here (you are too new to post links). FAQ default has 0 calls (max unlimited) in ‘ringall’ strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0. [from-intern - Trixbox Box :trunk iax2, sip extensions, the trunk is registrated ==> internal calls work fine but external calls the call is established, i hear the ring, but i dont hear the communication. I want certain extensions to be able to dial 911 and any other extension. 65-20. 1. The phone just goes back to the default screen (we are using GXP2000). When calls are sent to a cell phone the user wants to know that it came from “work”. End-user answers call and speaks/hears successfully. I am in the middle of that. My server is showing registration on Flowroute, however I am seeing multiple lines there where in the past, I only Hi. Its been suggested an expired Credit Card was used for something which caused the outage. Recently 2 of them got updated to version 14. I’ve set the outbound CID to the phone number they provided me. I can’t make calls . I tried to call and external number from ext 6000 and below is the logs pf the error Hi, I am a new to FreePBX and not to the Linux world. A few days ago, we had a power cut. They cannot make outbound local, long-distance or international calls. 7) allowed from anywhere and port 5060 forwarded only from two external IP address for two remote phones. I have followed guide to setup our firewall - PFsense as follows I have different port address for Media and it’s 6000-40000 and signalling it’s 5060. 10. Any suggestions please? I have a FreePBX server working and a new OpenVPN server. This normally happens around the lunch hour. I have configured static routes on CLI to route the needed Hi Everyone I’m Still having a lot of Issues with our FreePbx, actually we’re running 16. I checked the I have been running a few freepbx boxes with version 14. it always worked but recently it started that internal calls won’t work anymore. Here are my notes: Called the main number that rang into a Ring Group. The extensions are in the correct Pickup and Call groups as it works fine on external calls but drops internal calls after about 6 seconds. FXS extension: 3939 SIP extension: 5675 The extension is reachable when dialing from another SIP extension, however when I try to dial the other way to the SIP extension from the FXS extension, I am getting the below error: [2019-07-16 08:54:42] NOTICE[14516]: res_pjsip/pjsip_distributor. I was reading some other post on another community, and it said something about routing. We have applied the same template as all the other phones. to XXXX. for long distance call i have enbled the pinset by dialling zero infront of number. Set up a freepbx box (172. If a call comes from internal, the redirection to an external number (cell phone) works I’m using the FreePBX Distribution with CentOS and a AVM Fritcard 2. Sometimes, inbound calls work, but almost all are failing. When you call from outside, your truck is in use and appears to not be able to accept the second call over the trunk. x (the VPN client IP). The only difference between the two is the one that isn’t working is the destination of a Ring Group. Connected to Asterisk 13. I’ve tried restarting the Zulu server and my Zulu client Hello, I have configured digit manipulation rules for my second SBC to use a prefix for all outgoing calls: I’m doing automatic calls, by placing a file to /var/spool/asterisk/outgoing. I get the Call cannot be completed as dialed The provider has restrictions set to not allow cellphone calls on one and the other is not allowed to make landline calls. The incoming calls are landing fine but outgoing calls are not successful. 2 version of the elastix and tried to get the calls through Ring Group but it does not show the CID Name Prefix as it just show the caller number only and does not show caller’s name. 168. 4. We have recently changed over our ISPs and have a new network installed. Internal lines works fine. I’m using Telfree as my SIP Provider. Hello, I’ve put all of my internal extensions in a custom context, [from-internal-mine], which is defined in extensions_custom. I added my flowroute Using FreePBX 14 and the Cisco SPA504g phone, when my users dial outbound US number (e. We’re running FreePBX 13 (planning to upgrade to 14 eventually, but it’s a bear with Hyper-V) and use FollowMe with call confirm extensively. So the name and number are being stripped off at the PBX. My phones are all Aastra (6755i and 6757i) and I have a Sangoma A101 card. Also, we setup outbound route but it seems it’s not work. Hello, I’m working in a company where we use FreePBX as PBX. I’m running an SIP only build with Asterisk 11. Hi FreePBX users and admins, I’m running FreePBX with Asterisk 13. We successfully got outbound calls working but inbound calls still will not go through. I rarely use it, though. 1 with chan_capi drivers. I have setup port forwarding rules on PfSense for all the IP's needed listed here . 233 There I created an extension 200. If I call from another extension, the forwarding works. 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. The calls go through OK, complete with auto-answer on the headset or speakerphone, but they do not generate a CDR entry; whether they are answered or not, there’s no CDR. 18 running on a server (Centos 5 with Virtualmin), both installed using the repro’s. for example: When editing an extension i set the Outbound CID to “John Smith” <100> but in sip_additional. Does that seem plausible? I have two FreePBX systems (one Distro 13 and the other on CentOS running 13) I’ve setup two IAX2 trunks Internal PBX (Distro) Trunk Name: azsc-tie host=xxxx username=6301pbx secret=xxxx type=peer qualify=yes trunk=yes insecure=port,invite context=from-internal auth=md5 requirecalltoken=no Cloud PBX (CentOS FreePBX 13) Trunk Given that it says ‘extension not found’ it appears that FreePBX thinks my mobile number +3538948XXXXX is an extension, similar to my extension 201. The phone server is 192. Something my users need is the ability to call-forward from their office line to their cell phone. Internal Calls work 100%. the problem i am facing is Call recording is not working if i enable the pinset option. x. However, when I attempt to transfer the call on the MicroSip, the call is not transferred. The dial out in ring gorup look like this: 2000 2001 XXXXXXXXXX# (external This thread was a lot of help. Just bare with me. What steps to take to solve/investigate this issue? We have tried to add the prematuremedia=no property in the Hello, Since the upgrade to FreePBX 13, I have noticed that the CID Number alias is ignored for when calling out an outbound route with Intra-Company enabled. When we press a key on the phone, it’s not recognized. The phones are Place a call into the PBX from the phone number you previously specified while watching the CLI output. At certain point is My settings for extension Recording Options Inbound External Calls Yes Outbound External Calls Yes Inbound Internal Calls Yes Outbound Internal Calls Yes On Demand Recording Disable Record Priority Policy 10 but the call is not recorded, this happened after last update to Core 12. I have already done some tests with juste asterisk ( 13 LTS on a debian stretch ) and do some configurations to have some internal call, voicemail, and siptrunk. 210. When I receive the call on my cell phone which I have forwarded through Ring Group, it does Middle of the day today we suddenly stopped being able to call out of our building. The phone rings, I can dial out, but no audio either in or out. I have Ports 10000-2000 forwarded to server (192. What is working. I don’t have any prefixes setup and I am able to just dial the number which ultimately connects whoever I’m calling; in this case my cell phone. The direct call works fine for internal calls but we get a " app_dire After upgrading from 12 to 13 call recording playback in ALL browsers has quit working. 24 PBX Distro: 12. 9 PBX: v15. After long searching and testing a lot of things, I checked the modules. 0), Distro behind MikroTik router. We’ve looked at the extension settings and compared it to a working user with no differences found. I am trying to use the Custom Context Module (v 13. 58-x86_64-Full-1350436394) basically to act as a SIP gateway between a Lync Server and an old Cisco router. I have made entries for extensions, trunk (inbound/outbound), and outgoing route (with dial patterns and connected to the trunk) in FreePBX. 1 ). PBX Version: 15. I decided to randomly call my number one day and received a message “The number you have dialed is not a working number” I use SIP through Flowroute. makeapp the login succesed under “Trunks” but outgoing calls still not working. I am experencing this on 3 different If it is not working, then something is not right on your system. Hello, Firstly I would like to say that I have a basic knowledge of VoIP systems and asterisk/freepbx. creating a virtual extension Hi all, i am trying to use feature codes for attended transfer with the nice softphone tSIP (that i can strongly recommend everybody who dont know it as it is the only free softphone that brings customizable BLF-buttons! andeverything else we need!), but feature codes are not working on inbound calls. sng7 Asterisk Version: 13. I have installed even latest 2. Interestingly, when an external inbound call is coming or an external I was able to follow a PJSIP guide I found on Freepbx community forum and I’m able to get incoming calls. 4 and it seems that app-blacklist-check is not called at all in the dialplan when a call hits asterisk (at least in the from-internal context). Calls from one phone to another also work. Call rang until I hung hello, i would like to have a setup/call file check time of the day- if between specific time then proceed check a excel/csv/txt file for number to be dialed dial the number on connection , plays a specific message/audio file if person wants to connect further, then on pressing the key, connects to preassigned number (ext/trunk call) - optional if person does not want to connect Line 1 and Line 2 have two different providers; Line 1, (US provider) can call the US (local and long distance) and receive calls from anywhere. I do have some xp with OS’s and programing, but I’m completely newbie with VoIP. 3 and the default gateway/firewall is a CentOS 6. 25(13. External calls are working perfectly. What I can’t get to work is internal calls from a device on one gateway to a device on another gateway. When I tried Yealink or Sangoma phones the transfer appears to work fine. The auto-setup put my FQDN in the domains as allowed. conf file) the caller ID will not show up correctly when calling internally. 23. 0 and FreePBX 13. Is there something i need to know about FreePBX 12 with regards to audio issues on Ive been battling to set up my pbx for about a week now with Twilio as my SIP provider, comcast as my home network ISP provider and freepbx running on a VM inside Proxmox, aswell as PfSense being my main router. Outbound Calls Are not working. Calling out stopped working while everyone was out to lunch it seems. The phone is registered to me in Sangoma Portal and assigned for “Enable Redirection” with redirection type as Deployment (I have also tried the Latest update: As per my recent call with the provider, they are not receiving any request from the Freepbx server. When a user sets any or all of the three call forwarding conditions (Unconditional, Unavailable, or Busy), and you Hi guys, I currently have some problems with a forwarding which is set up on the phone (Yealink). c:649 FreePBX. Only incoming calls are not working. (Analog, PRI, SIP) If the call is reaching your PBX, verify internal call handling. exe) on 88888 Hi All I need a little help here. ) If I call from another Hello everyone, Thank you very much for the great product and for all the advice available from the community. Configuration. We had a fully configured PBX 100 and switched it over, and I have a Digium D40 phone set-up at a satellite office as an external SIP extension. 18 FreePBX 2. 19. But I cant get outgoing calls to work, it must be something simple like a port that needs to be opened, can anyone offer a common outgoing call issue. It’s the only extension at that location. The people that Just getting started in setting up FreePBX. Not remote phone. I’ve had a problem with dialing extensions in the past, and it usually goes straight to voicemail, but this time it does not even We successfully got outbound calls working but inbound calls still Hello all, We switched to a new SIP Trunk provider however we can’t get inbound calls working. I need to divert calls using ring group placing the mobile number with a “#” at the end along with one of the extensions. However, starting today every single time when we hit 1 the PBX just ignores it. 11. [C-00000024] pbx. I am a 25-year technical person (software developer), but brand UPDATE: Call monitor under voicemail and recordigs also does not show any call history or recordings even though /var/spoo/asterisk/monitor/ shows recorded calls. There is no SIP ALG option on the remote router. Could any of you give me some tips to resolve this very ext201 - works fine ext219 - does not. I have created 5 “Misc destinations” with 5 mobiles numbers, and then 5 virtual extensiones 10001-10005, those “Virtual extensions” have the “no answer” section to the “Misc Destinations” Now i have created a Queue (600) with the 5 static agents, from 10001,0 The direct call works fine for internal calls but we get a " app_dire Hi, we have a FreePBX Distro 4. There are only 2 lines in this context: [from-internal-mine] exten => _[*#0-9]. Example, 1 for Customer Service, 2 for hi i am using asterisknow with freepbx2. I was hoping to leverage FreePBX to record all calls passing through. This 1900-01-02 is the default date shown under this circumstance. 22. There is no tone as the phone Hi, Not sure if this is a FreePBX or Sangoma issue I have a bunch of S700 and S500 phones on a local FreePBX machine with both SIB and DAHDI trunks. **All 18 endpoints ring: but can only answer if they answer on the 1st ring only Then: using module admin; We updated the ring group module to: 15. Good morning. The phone works as expected to place outgoing calls and when used for internal calls. 64-5 working fine except for the Direct Call pickup of external calls. I am currently on the latest build - 10. I have mine setup, incoming calls go via my IVR, calls are flagged based on incoming caller selection, voice mail works a treat and I have 2 VoIP IP phones in the office both connected. If the person at the remote location It appears that a call that is originally answered from a Ring Group call is affected. We are having an issue with a single remote extension at a location. 0 Brand new P330 trying to set it up with PBXAct internally. 2. 17. 28. I have asterisk freepbx 13. Is this possible? I figured out how to remove the CID for internal inbound calls, but was not able to do a similar context for external inbound calls. All phones and PBX are on the same switch and network. I have a couple of SIP accounts on a couple of android phones. 4 (all updated) and have not been able to figure out how to get group paging working. Calling other extensions works just fine. It appears to be in FreePBX as it is not leaving our ne Using the SNG7-PBX-64bit-1910, I am trying to get outbound routes working with my new Clearfly SIP Trunk. I forwarded RTP ports 10,000-20,000 to the phone, it still did not work. 711. Outbound calls work, but inbound I Sangoma PBXAct 25 LATEST firmware/everything: PBX Version: 16. Hello people, I know that this topic has been discussed a lot, but not for FreePBX 16. Please check log below. I have done the following so far: -installed the paging and intercom -installed the endpoint manager - i configured the phone using endpoint I’ve set up freepbx 2. I assumed the User Control Panel would let them easily do this, but it doesn’t seem to be working. If I make an internal call from the main office where the server is to this remote location I can not hear the person at the remote location and they can not hear me. Hello I’ve tried setting up a blacklist number for: John Doe 1-234-567-8910 In my Blacklist module I’ve added phone number 1-234-567-8910. My suggestion is try to give us more details how to @david55 suggested you. Went from Comcast to ATT, and from a 192. I have SIP lines online under Asterisk logs. Help, please. I am relatively new to FreePBX. I haven’t set up Hi @Si_K My understanding, yours Outbound calls from Zoiper (softphone) calls are not working to out. 106 using a Raspberry Pi 3 and Yealink phones. FreePBX Hi All, I am having an issue where all internal calls are ringing 2 times and going to voicemail. Awesome. from x-lite ext201 - works fine ext219 - does not. 04 I have enabled “CDR Logging” in the Advanced settings, rebooted, and “cdr show status” shows: cdr show status Call Detail Record (CDR) settings Logging: Enabled Mode: Simple Log calls by default: Yes Log unanswered calls: No Log I have a freepbx installation using the internal firewall and fail2ban right up against the static external IP address. Hi, I’m currently in the testing stage of our migrated on prem server to cloud but we have serious issue when the extensions (zoiper use for testing w/ external IP) receive calls, the audio and microphone are not working, I have installed FreePBX Distro (FreePBX-1. This is where I expect SIP registrations for my softphones to occur. When a caller dials one of my extensions from outside, they hear silence for about 2 minutes. 66-16. The CallerID Prefixes in FreePBX are more for “internal” calls so you can see where a call is being sent from like a queue, ring group, etc to help CDR doesnt work on a fresh install of Asterisk 20 and FreePBX 16, manually installed, on Ubuntu LTS 22. de website (thank you google translate!) FYI for people looking in the future: pjsip Settings in the General tab for the trunk: Username : sipgate user name Secret : Sipgate passowrd Authentication : Outbound Registration : Send Lang code : English Sip Server : Hello, I am currently using FreePBX 12. I have a firewall and have opened ports 10000 to 20000 and 15000 to 30000 along with 5060 (udp and Outgoing calls works too, but not incomming (caller [me trying to ring in] is just disconnected). All my extensions are working fine on internal calls, i can reach my pbx from ouside, but i cannot place calls. I have tried We have several remote offices with remote extensions. 1 running Asterisk 14. 5 as a part of asteriskNOW, everything is working okay except internal call ID, no matter what i do (bar modifing the sip_additional. The SIP is registered and incoming calls are working properly whereas the outbound calls are failing with All Circuits are busy now, please try your call later; This seems to be a firewall issue, but after careful examination, I still have not identified the exact problem especially since some external calls work and others do not. so, if somebody calls our FreePBX and i answer the call with tSIP, Hi, on a FreePBX 2. 58. Next I enable “Block Unknown/Blocked CallerID” and terminate We have a system that is having a weird issue with PJSIP extensions. Here are all the modules running. I can call between the 4 digit extensions just fine. 8 to Asterisk 11 and then upgraded the Freepbx Distro from FreePBX Distro 5. 212. I made all the latest updates, and I forced the reinstallation of the IVR module with the latest version. But there was none. One interface (eth1) has the default gateway and is facing my local network. (Time Conditions, Queues, IVR's) Merging calls in Zulu does not work for me. External to IVR selection. But call recording is working for local calls for This happens only if I call from an external number. 5. The other interface (eth0) faces my SIP provider. It started on 5/21/24 with no known changes to the PBX or ISP. Before the call is dropped, there are no problems with audio. However, when the dial 555 for an I am able to make calls out from the phones but no calls in work and I get a Request timeout with sip when trying to call in. Paste the complete Asterisk log for the call at pastebin. I have recently installed a FreePBX instance with two NICs. Applications / Modules. 3 and o2. 0 I just installed from the 1701-1 ISO. Calls that work, I see traffic in My PBX had previously been working just fine with no issues. 72 ( Current Asterisk Version: 16. ,1,Set(TIMEOUT(absolute)=7200) same => n,Goto(from-internal,${EXTEN},1) As you can see, the above context simply sets a time limit for the call This used to work but recently stopped working. I then try to make an outbound (both calls to 701. 5-1807-1. I’m using (what was) the latest stable version - FreePBX 2. With a tcpdump at FreePBX server i see packets coming from 192. 42, we just had the Problem with any Type of calls Internal and external. 1) on our FreePBX (v 15. 8. . Having trouble with call forwarding to external numbers (e. 1 and on the other I have 2. Outbound calls are getting a message “the number you have dialed has not been recognized”. The recipient phone does not ring at all. I see the IP of my cellphone registered in the firewall under Successfully Registered Endpoints so I don’t think it’s a firewall issue? Anyone have any ideas? It works fine when on my LAN After upgrading from 12 to 13 call recording playback in ALL browsers has quit working. I have watched asterisk when I place the calls. Any ideas or thoughts would be great. Please see the logs for Hello, This just started happening and was working. Nothing changed on the firewall at all during the I have FPBX installed, got 26 x 24 port Cisco VG224 analogue gateways attached, pjsip trunks to each gateway, inbound route for each gateway from PSTN, outbound route to PSTN, and all lines are working fine for PSTN in/out calls. I am able to dial from one extension to the other, but cannot hear voice from either extensions. 2 and other modules. BLFs do update showing that the phone is ringing, even though it is not. system (system) Closed May 29, 2022, 8:15pm Hello everyone, I have configured a freepbx on my local pc, i’m receiving some issue with inbound calls. The issue is that randomly a phone will become an unknown device and as such asterisk recognises the phone as anonymous so will not allow any internal or external calls to be made from the phone. c: – Executing [077616102@from-internal:7] [/color] Macro [color=#ffffff; font-family: ‘Courier New’, Courier, monospace When we make outgoing calls we don’t hear anything until the other person is connected. (I also tried Lorne’s pay-it-fwd script to bypass the Cisco hardware, but get the same results. , kindly guide how to fix as in GUI no such option was there The call is getting connected to the SIP server in private We have a small office that is using FreePBX and linksys IP phones, I recently added two new devices and extensions but when you call the extension you get a message saying “the call cannot be completed as dialed” or to that extent. My extensions are 10-digits (local phone numbers) and there is a matching dial pattern in Are you running a distribution that came with Freepbx (if so which one) or did you build it yourself? Do inbound calls to extensions work? Have you looked in /var/log/asterisk/full You are not the only one having problems. 2 currently running on localhost (pid = 1835) – Added contact Hi: Any tips for fixing this? FPBX has a SIP trunk (working) Incoming calls are set to be answered by a ring group (with about 18 endpoints) The calling party gets NO ringing sound when calling in. calling out and in from sip account is working fine when I setup with the softphone on my PC. all the internal calls work; i can make calls out and receive calls in however I cannot hear any sound from the external call. All modules are succesfully loaded to asterisk. Internal extension works fine. 1 on an ubuntu 8. Freepbx logs the call with Destination “hang”. Latest update: As per my recent call with the provider, they are not receiving any request from the Freepbx server. Since then, IVR are not working anymore. sng7 Asterisk Version: 16. Thanks. It has always worked fine, but since a few days not anymore. This is what is funny, I have it so the phone is making a ringing sound, but the number I called Adding a prefix to internal calls is working but not when using external number in follow me list. It is strange behavior. 18004377950 , paste the complete Asterisk log for the call (which will include a SIP trace) at pastebin. I have taken it off just the phone, then I just took it off then extension setup and finally I took it off both. 1 with Asterisk 1. 6 calls from one extension to another were internal (free) calls. Thanks CSeq: 102 OPTIONS Call Having a problem with routing calls to my mobile after hours or after I leave early. 211. The external SIP phone will ring, but when the receiver The normal business hours IVR will not accept DTMF (from any source) and the after hours IVR works fine. cell phone numbers) not working. 65-13 to 6. Blocking extension to extension did not work for me though. All direct ring in DID calls and direct ring in DID calls that are transferred work. Resources Firmwares, tools and documents. 16 Before the I have a SIP Trunk configured and is registered in FreePBX. I’ve setup freePBX with 3x Softphones and 2x Snom phones connected to it. Outgoing, incoming, and internal calls are now working fine it seems, but if I set my inbound destination to IVR, an incoming caller just hears “good bye”. I have a fresh new install of FreePBX 16. 100 after that the caller will get a message “Check the number and Dial again” ? Our parking lot is being resurfaced & the people in one building are working out of the other. After I attempt a transfer I still send and receive audio on the call, however after about 30 seconds I run into the Request Timeout Issue and exten => _XXX,n(notfromext),Goto(from-internal,${EXTEN},2) The above actually works in testing, but it’s not right. 34 I’ve looked through a handful of related posts and am drawing a blank. I didn’t change anything hadn’t even logged into the pbx in months. Incoming calls are working just fine. 4 I’m having troubles setting the “default” MoH for internal calls (extension<->extension) and calls placed on park. Observation: All internal calls on the LAN have no audio issues. No configuration changes were made. But, when an inbound call comes in from an outside phone the external SIP extension does not answer the call without a workaround. can anyone help me solve the following problem? I am new to freepbx and i am having a problem with getting the intercom to auto answer. Others may read the log better than I. FreePBX. Of course if the user’s follow-me is disabled then it will skip the follow-me settings. x but from asterisk console i see that the extension address is 10. I use softphones (phoner. 2 on Asterisk (Ver. Internal calls to other remote offices work fine. When I try to ring another extension connected to FreePBX, it works. I created a new context in extensions_custom. 3. , 15555551212) the calls automatically dial. Many Thanks. 5263) and it failsI then make an internal call using Hello, I’m trying to forward calls after work time with “Time Conditions” but it don’t work to me here I’m gonna post the pictures of my configuration This is Misc Destination of my Cell phone Here you got the “Time Condition” This is “Follow Me” configuration: I got 2 rings on ext. Call forwarding to internal extension works. 16. , because previously these internal calls would be routed out via a trunk; tried calling both 101 and (with In trixbox 2. The issue I am having is that after I dial my outgoing number (my cell phone), I just hear silence. 8-2208-2. conf called [From-Internal-new] and added that context to a test extension. I have tried every variation of settings to record all calls on my Hello, I am new here, and i am not familiar with asterisk and freepbx, i’ve just started a project that consist of the deployment of a telephony solution inside an aircraft, with SIP trunking to allow calls from and to ground. Whilst caller ID is being passed correctly from both trunks, the name lookup from Contact Manager within FreePBX is not working. I created a Custom Context “EmergencyOnly” wherein I allowed only “EMERGENCY” on Outbound Routes, while the rest of the parameters I am running FreePBX 14. Target IP: <Internal IP of my PBX behind pfSense> Redir. I’ve tried everything I can find via google but nothing seems to work. I can call out from cisco but when I try to call from asterisk phone out the call does not work. When I ring the DID associated with one of the extensions, it works, so inbound calling is working. c:670 remove_header: No headers had been previously added to this session. Greetings, I’m not able to get my NAT settings and firewall setup correctly. 12. 817. The issue is a lack of audio on PJSIP extensions on internal calls when connected from some public IP addresses. Another thing I’ve noticed is, that when I use the phone’s forward function, when I call from another extension to the one which i want to be forwarded, the caller ID shown on the external phone is not correct. freepbx. 4 and after that the outbound route CID has stopped working. It's only when an outbound calls are made or an inbound call is received from an external source there's no audio. Eventually a telephone company recording saying “I’m sorry your call did not go through please try again”. Calls into the extension or out of it send and receive audio fine. WHAT WORKS: Internal extension to extension. I have two outbound trunks for Hi, I have the following setup PBX Station Aastra MXOne as main phone station. This setup works as follows: Outside Extensions This will include Direct Calls, Ring Group Calls, Queue Based Calls, IVR transferred calls, etc. You said it works when you call from an extension but your trunk isn’t in use at that time. 0% within 0s FreePBX DAHDI setup stuff should work, we’ve seen correspondents galore report that they had it working, then tried to check the config with the FreePBX stuff, only to find that it no longer worked. - I can make outbound calls from any of the extensions, (sweet) - I can call into the external DID on FreePBX, that works fine until FreePBX routes the call to any of my extensions ( fails with "No Response" ) Redir. Since then We are receiving some errors when dialing outbound. The cause appears to be the “retransmission timeout reached on It’s not finding a trunk available to place the outside call. 1 Like. Outbound calls are working, no changes have been made and this was working yesterday. 1 running on Raspberry Pi 4 and everything is stable. Any Suggestions would be greatly appreciated, I started a post in the trixbox forums but not getting to far. PS: I do not currently have any SIP trunks associated to the FreePBX, so can not test “real” calls, but I assume that if internal calls don’t work then other ones won’t either. Internal line to Line calls work No access currently to the linux box its running on. Does a c. Even though on the Outbound routes I have a dial pattern to prepend 1 on numbers from The “Free” in FreePBX stands for Freedom. Now I can receive internal and external calls and can also make calls to extensions Hello. but when I move that sip account to used with Asterisk and freepbx have some problem with outbound call. Interestingly, when an I managed to get it working once I found some extra configuration information on the sipgate. all i hear “all circuites are busy now” here is a verbose of the asterisk console if anyone help me. In all this months I never experienced a problem, no downtime, no apparent intrusion attempt (I allow only Hi Everyone, For some reason, my system lost all audio for inbound calls. But with any I began setting up a FreePBX install some time ago and today have spent sometime getting it working with SIPGATE here in the UK. When i watch it on the CLI, i always get’s kicked out of it like that: 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. Extensions register on the internal profile, gateway registers on external profile (port 5080). 13. conf. 29 with Asterisk 11. JNBTECH (Tech Team) November 9, 2021, 3:18am 1. Target Port: SIP If I call the office’s inbound 10-digit phone number the audio works fine, it’s just internal calls from one extension to another. Looking again I see it’s moved on a bit, I’ve been at it for a while between other things. For some reason, internal calls do not work. Extension Settings: Outbound route: Is there something else I must do or enable to get this working? Internal calls within the same system show the CID Alias number. The log Mic/Audio not working from external/internal calls. I have so far: checked that both extens use the context from-internal; changed my outbound route dialplan from X. This has worked fine in the past, where every now and again when we hit 1 to confirm the call it doesn’t take it. Here is my trunk configuration: OUTGOING SETTINGS: Trunk Name [Telfree49] Hello, I run FreePBX within Elastix 2. 8 with Asterisk 16. The issue is I only have one trunk for both lines and its set as round Hi My CID Name Prefix is not working in Ring Group Module. 2+215 Zulu Server: 15. My IVR is very basic at this point, it’s supposed to play an announcement and loop twice. 0. 1 i don’ have it working and cannot seem to get it to work. I have the logs pasted below: Hi All I have an issue with call pickup (*8) on Asterisk. I have encountered an odd issue with a new pbx setup. Nothing happens when I dial the page number. I can’t dial out, I can get incoming calls, FreePBX registers with Flowroute Can somos one please me help me out? Like I said before incoming calls work it’s just the outbound are not working. 8, Call Recording 12. The problem here is returning to the from-internal context. I can make external calls via my SIP without any problem, but I can’t receive any incoming calls from the outside. cbftlj fsetax kszm tgask mxbpg vznbj zhe hkd doohd ppn