We have collected the most relevant information on Freepbx Audio Ports. Open the URLs, which are collected below, and you will find all the info you are interested in.
Configuring your PBX or device with SIPStation ... - FreePBX
https://wiki.freepbx.org/display/ST/Configuring+your+PBX+or+device+with+SIPStation+Service#:~:text=We%20recommend%20forwarding%20ports%20UDP%2F5060%20and%20UDP%2F10000-20000%20for,SIP%20Settings%20for%20the%20bind%20port%20of%20ChanSIP.
Ports used on your PBX - PBX Platforms - FreePBX
https://wiki.freepbx.org/display/PPS/Ports+used+on+your+PBX
Zulu 2.0 requires this and the ports below to be opened. NOTE: You may require the "RTP for SIP" port range to be open as well, for call audio. 8088: TCP: Zulu 2.0 Unencrypted Softphone Client: Can change this port inside the PBX Admin GUI > Advanced Settings > Asterisk Builtin mini-HTTP section > HTTP Bind Port
No audio on inbound or outbound calls - General Help ...
https://community.freepbx.org/t/no-audio-on-inbound-or-outbound-calls/47303
I am running FreePBX with Asterisk version 15.1.5. I am using an old ObiHai 110 device as an FXO port and a Gigaset C530IP DEC station as a PJSIP extension. Both devices register with PBX and calls can be made and received but there is no audio in either direction. When calling the extension’s voicemail, the logs show that the proper audio files are played by …
Configuring your PBX or device with SIPStation ... - FreePBX
https://wiki.freepbx.org/display/ST/Configuring+your+PBX+or+device+with+SIPStation+Service
We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk-based installs. If using newer versions of FreePBX, port 5160 is the default port for ChanSIP so that may be the port you need to forward. Check Asterisk SIP Settings for the bind port of ChanSIP.
No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/77854
No Audio PJSIP. FreePBX. Endpoints. carminuch (Carminuch) 2021-09-08 09:15:06 UTC #1. Hello all, I currently have 2 VM’s set up for calling each other, and they do so just fine. However, once they connect there is no audio. They are calling each other over PJSIP, and both are capable of doing the echo test.
Softphone Audio Troubleshooting - Zulu UC - FreePBX
https://wiki.freepbx.org/display/ZU/Softphone+Audio+Troubleshooting
If your PBX is behind a firewall and your Softphone is not on the same LAN as your PBX you will need to open the Zulu Softphone Port on your firewall to your PBX. The default port that Zulu talks to your PBX for the Softphone is Port 8089 using TCP. Verify Windows Firewall is allowing Zulu access Debugging Audio Issues
SOLVED: No audio on remote extension - General Help ...
https://community.freepbx.org/t/solved-no-audio-on-remote-extension/38501
Hi, after replacing (an old) Freebpx installation with 13, the remote extensions are able to register, intitiate calls, but there is no audio. During the call setup I can see a 401 error, and after a few seconds the line is dropped because no response from the external extension. We are running a NAT setup, no SIP ALG, same NAT setting as the old freepbx system. I’m sure it’s …
Distro and PJSIP: No Audio - FreePBX - FreePBX Community ...
https://community.freepbx.org/t/distro-and-pjsip-no-audio/52675
To test chan_sipm I also set up an extension for chan_sip on port 5160. chan_sip extension audio was functioning as expected, but for PJSIP, no audio could be heard. This is setting up new Sangoma OS w/FreePBX 14 three times now. After swapping the default ports with chan_sip to port 5060, and PJSIP to 5160, I am now getting audio.
No Audio until resumed from Hold (remote) - FreePBX
https://community.freepbx.org/t/fpbx15-pjsip-no-audio-until-resumed-from-hold-remote/69556
Hi, Need some help, first time deploying freepbx 15 and also pjsip. Everything works okay internally but remote phones are causing me an issue. There is no audio either direction from the remote phone until you place the call on hold and resume it. Then it works fine. This is usually a NAT issue somewhere but with PJSIP I don’t have any NAT options like with …
Ports used on your PBX - FreePBX
https://wiki.freepbx.org/spaces/flyingpdf/pdfpageexport.action?pageId=5538035
Standard Port used for chan_PJSIP Signalling. 5061 € chan_PJSIP Secure Signaling Can change this port inside the PBX Admin GUI SIP Settings module. Not recommended to open this up to untrusted networks. Secure Port used for chan_PJSIP Signalling. 5160 UDP chan_SIP Signaling Can change this port inside the PBX Admin GUI SIP Settings module.
Now you know Freepbx Audio Ports
Now that you know Freepbx Audio Ports, we suggest that you familiarize yourself with information on similar questions.
