sip

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...