sip

Hi, I have a gateway that works for inbound calls. Outbound is not working as the SIP carrier requires the INVITE, FROM and TO, to have the domain of the SIP carrier, but its using the proxy address for the INVITE and TO header portion of the trace. The Gateway has Domain = realm set ,and proxy...

Hi, Is it possible to change the default internal SIP port (5060) to another port? If yes, could you please advise how to configure the softphones to connect to FusionPBX using the new port? The objective is to free up port 5060 for use with the external profile. Thank you.

Hello everyone, I’m currently using a Yeastar TA410 to connect an analog line (FXO) to my FusionPBX system. Outbound calls through the TA410 are working fine, but inbound calls are not reaching FusionPBX. Here is my setup: TA410 FXO connected to the PSTN line. TA410 SIP trunk to FusionPBX...

We are experiencing a common, yet problematic, SIP dialog flow between our Freeswitch (v1.10.10) and the Zoom SBC, which frequently results in hanging calls. We are seeking insights into why Freeswitch fails to terminate the session properly on the first attempt. The Problematic Call Flow...

Does anyone have recommendations for SIP client phones: (Grandstream, Yealink, ....) ? Specific models that work well with FSPBX. Thanks.

I've reinstalled a new FusionPBX since I'm still running a very old implementation of this PBX solution. I've copied the configs a tried to do a switch of the PBX systems to the same IP the original old 3.x version had and phones are connecting with. Now the issue is, that on the new PBX...

OS Working on : Debian 12 AWS EC2 Machine Instance with Public and Private IP Extensions are Setup

Hello fellow FusionPBX-ers. I am near to pulling my hair out on this one... I have Yealink T46S phones. It's a commonly-reported problem that when you go in to DND mode, the line key LED will not switch to red on everyone else's phones for your extension. So I thought I could spoof the NOTIFY...

I'm working on a POC using LibreSBC which uses freeswitch and when sending test SIP INVITEs with the StarTrinity client and including the 'require: siprec' header, I immediately get a SIP 420 Bad extension with Unsupported: siprec In...

I am using FusionPBX as SIP server(registar). FusionPBX works fine for call and SIMPLE message routing, but I need to modify it so that I can send SIP/SIMPLE messages to registered users from FusionPBX. I found on freeswitch docs site that it can be achieved from lua scripts or xml files. Here...

I need to use FusionPBX as SIP registar(Session Initiation Protocol) or SIP server, to route SIP calls to appropriate SIP accounts (extensions). Currently every extension has possibility to call any registered extensions and everything works fine. But for my purposes I need to set some logic...

Hello Hello everyone as I was trying to configure my SIP I can go out but I can't go in if I remove the provider from the ACL it says Auth is required when I add it, it says 480 Temp Unav. and i attached a log with the last 150 KB and a screenshoot of sngrep with the request

Heads up. Sofia SIP can possibly bring down your PBXs with an easily craftable OPTIONS message. https://github.com/freeswitch/sofia-sip/security/advisories/GHSA-h94r-c3pv-4564

I can't seem to find any documentation to raise the max-forwards from 6. Any help would be appreciated.

Hi everyone, Hope you're doing well. Suddenly, on Bria app I get SIP error 408 and no inbound calls are not not working. When I checked the logs, it has only one line which says: mod_logfile.c:192 New log started. So I guess there's something which is blocking every requests. I checked the...

We have a FusionPBX on a VM in the cloud with a public IP we use Bandwidth.com as our SIP Trunking Provider (We are IP ACLed and do not use REGISTRATION) We have several clients at different location with several phones each that connect to our phone system in the cloud Everything works great...

Hello, I have a fusionpbx with video call enabled, Since last Thursday the video calling got stopped automatically, I have attached a video call log, I tried to Debug this a lot but could not find the problem. Can anyone please help me out ?

hi, I have an asterisk based telephony system running on centos 6.x ( esxi ) , there are IP PBX, mediant on the system. there is a problem on calls, calls drop after a while, cli says restransmission, provider warning, some of user blocked by the system , system runs sometimes slowly too, we...

Hello, There are many SIP clients available for mobile phones and many of them work very well with "simple" SIP trunks but when we move into the PBX arena the mobile app needs to get more sophisticated handling presence and Busy lamp fields etc. What are people using for mobile phones...