1. R

    Cannot start Freeswitch - Help!

    Hello all I hope someone can help quickly. Here is story: My FS/Fusionpbx server ran out of disk space. The postgres server stopped responding. I quickly removed a bunch or files to make space. I restarted postgres. I also restarted Freeswitch using service freeswitch restart - and got an...
  2. hfoster

    SOLVED Multiple registered endpoints, one breaks the forked invites by sending 488 Not Acceptable Here.

    Wonder if anyone knows the secret FreeSWITCH magic to get FusionPBX to ignore a 488 Not Acceptable Here error message from a single registered device (in this case a PUSH server) from breaking the entire call? I've included a diagram of what's happening to the call. Normally, the Push Service...
  3. M

    invalid target uri in NOTIFY - missing "sip:" prefix

    FusionPBX Version: 4.5.27 Switch Version: 1.10.7 (64bit) OS: Debian 10 Hi all, I'm facing a really strange problem with invalid NOTIFY messages. More precisely, it's about the target URI in the xml body. It sometimes happens that freeswitch does not send a "sip:" prefix there, which...
  4. D

    Outbound route 483 too many hops

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

    SOLVED Getting "Connection to Event Socket Failed"

    Hello All I am hoping someone can help me quickly. I am getting the above error whenever I go to Reg, Calls, Sip Status etc. I had changed the event password (in Settings) from the default and it all went wrong. ( i have since changed it back to the default) I have restarted freeswitch many...
  6. D


    Hello I have a issue with my inbound calls becausa are denied by my server. I response with SIP 403 Forbidden. This my internal configuration profile (internal and external has the same configuration with autonat:public_ip). The issue is only with the inbound calls, outbound call work...
  7. K

    Calls Get Stuck in Call Center Queue Randomly

    PBX version 4.5.32 FreeSwitch Version 1.10.6 (64bit) We use the Call Center feature with about 20 agents and 5 inbound numbers. We get around 20-30 calls an hour on average. Randomly calls get "stuck" in the queue - the call ends, but the call still shows as active in the Active Call Center...
  8. S

    Routing message through gateway

    Hi, I am new to FreeSWITCH and I am trying to send a message from Freeswitch-A to Freeswitch-B. I have created a gateway connection. Registered one softphone 101 with Freeswitch-A and 6010 with Freeswitch-B. But when I send a message to 6010, it says can't find registered user...
  9. T

    ESL and Timeout

    Hi! I am using the C# NEventSocket library to dial some some numbers. I have some Tasks that create an InBoundSocket, connect to them and send the originate command. It works most of the times, but sometimes it throws a timeout error: NEventSocket.InboundSocketConnectionFailedException: Timeout...
  10. F

    Bria SIP error 408

    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...
  11. V

    Outbound call from internal extension is rejected

    Hello, Im new in this forum, and i thank you in advance for your help. I create and configure a fusionpbx just with a public IP (without domains) and create extention "20". When i run test call to another extention "21", i have a rejected call. Can you help please ? 2021-08-17 16:09:58.153982...
  12. F

    Test-your-call service

    Hi all. I want to share Test-your-call. This is a web page that contains some DID numbers for testing and debugging audio on VoIP calls. Link:
  13. S

    Change cipher TLS

    Hello, This week I'm trying to configure FusionPBX with Homer and Captagent, but Captagent keeps giving errors because the DH cipher, which is not supported (ofcourse, is used for TLS encryption. So I looked up what cipher(s) are supported, and those are these two...
  14. E

    Freeswitch issues with flutter dart sip ua

    Hey there everyone. This is probably the wrong place, but I wondered whether anyone could still help. I'm not using FusionPBX, but a merely default configured FS, with Flutter Dart SIP UA. I'm having issues with setting this up correctly apparently, because I can't receive any calls on the...
  15. I

    Dahdi AX400p

    Anyone here have tried adding dahdi hardware analog ax400p on their fusionpbx box? any suggestion and idea please care to share Thank you
  16. I

    Modify Contact header FreeSWITCH

    Hello, Does anyone know how to rewrite contact header on B leg call of freeswitch? on bridge statement by default its contain "". just stuck on sip_contact_host value, its still contain ip address value. expect value looks like "Contact: <>"
  17. K

    Fusionpbx Install on existing freeswitch

    Hi All, i want to know if fusionpbx can be installed on existing working freeswitch. as per install docs, Fusionpbx installs freeswitch with it. i want it otherwise. basically, i have working freeswitch but need GUI interface for users to control Conferences like look at active conference...
  18. M

    Forward / routing loops, full database connections

    Hello, I'm using Fusionpbx 4.4 and struggle a bit to get call groups setup properly. I have a callgroup with ext. no. 100 and some members. If some members are forwarding their direct calls to ext. 100, they get called by the group again and redirecting again too the group. Is there a way to...
  19. A

    Unable to configure outbound calls to PSTN

    Hi I have configured FusionPBX on ubuntu machine with two network interfaces - One interface is connected to WAN/Office LAN and other is connected to SIP LAN network. I created new internal SIP profile so that the freeswitch makes connection to both my network interfaces. I am able to receive...
  20. H

    High load average with low CCU

    Hi everyone, I'm using FreeSWITCH-1.6.20~64bit on Linux. We tested successfully on dev enviroment with 3000 CCU without any problem, but got issue on product server at 500 CCU. - The scenario is very simple: + sipp send INVITE at 3000 CCU, + freeswitch play audio + after 10s, sipp send BYE...