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Asterisk behind pfsense (no sound) | Netgate Forum
https://forum.netgate.com/topic/37420/asterisk-behind-pfsense-no-sound
You redirected ports from 17000 to 18000 to your sip server. Your second sip device received a call and remote server sent rtp to 17454. In this case you will have no audio and pfSense is not guilty. SIP issue sample: You redirected port 5060 to your sip server. Your second sip device registers at voip.com.
r/PFSENSE - SIP Phone no audio. Rule set to allow all ...
https://www.reddit.com/r/PFSENSE/comments/8r2xj8/sip_phone_no_audio_rule_set_to_allow_all_traffic/
The phone system is hosted in a data centre which is fronted by pfSense. There are lots of users connecting to this phone system without any issue and they all get two way audio. My local network is protected by a local installation of pfSense. Within the LAN I have a local test phone system which has SIP trunks connected and they are working fine.
Asterisk/FreePBX behind pfSense – no audio in/out – iTecTec
https://itectec.com/superuser/asterisk-freepbx-behind-pfsense-no-audio-in-out/
Asterisk/FreePBX behind pfSense – no audio in/out. ISP modem in bridge mode -> pfSense firewall -> HP2920 switch -> asterisk | VoIP phones. I finally got inbound and outbound calls working but I hear no audio in/out. If I call the phones internally, I hear both sides. pfSense's NAT port forward is set to any/any for IPv4.
PFSense Firewall Settings for VoIP – OnSIP Support
https://support.onsip.com/hc/en-us/articles/204029430-PFSense-Firewall-Settings-for-VoIP
Disable source port rewriting - by default, pfSense rewrites the source port on all outbound traffic. This is necessary for proper NAT in some circumstances such as having multiple SIP phones behind a single public IP registering to a single external PBX. With a minority of providers, rewriting the source port of RTP can cause one way audio.
SIP - No audio or one way audio :: Zoiper
https://www.zoiper.com/en/support/answer/for/android/11/SIP_-_No_audio_or_one_way_audio
SIP - No audio or one way audio ( on Android) « Back. When using SIP protocol one way or missing auido issues mostly appear due to configuration problems. The issues usually appear if the configuration is not adjusted accordingly to the wireless or 3g/4g network. In most cases this can be resolved by altering the account configuration. ...
pfSense Configuration Recipes — Configuring NAT for …
https://docs.netgate.com/pfsense/en/latest/recipes/nat-voip-phones.html
By default pfSense® software rewrites the source port on all outbound traffic. This is necessary for proper NAT in some circumstances such as having multiple SIP phones behind a single public IP registering to a single external PBX. With a minority of providers, rewriting the source port of RTP can cause one way audio.
SIP problems behind pfsense box : PFSENSE
https://www.reddit.com/r/PFSENSE/comments/31a1y1/sip_problems_behind_pfsense_box/
Hi guys! I've recently decided to move to a pfsense box from an ASUS RT-AC68U router, but the issue seems to be that the pfsense box is being overly restrictive and blocking my Asterisk box, whereas the Asus doesn't. With the pfsense router, two out of three sip trunks are unable to register, but all are OK on the Asus router.
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 …
pfSense Configuration Recipes — Configuring NAT for a VoIP ...
https://docs.netgate.com/pfsense/en/latest/recipes/nat-voip-pbx.html
pfSense Configuration Recipes. ... for the upstream SIP trunk addresses, if known. If the SIP_Trunk address/network is not known or changes, do not make an alias and leave these values set ... Once the PBX re-registers it test inbound and outbound calls and confirm inbound and outbound audio works as expected. Next Configuring NAT for VoIP Phones.
PFSense Firewall Freevoice SIP Phones - Admin Guide ...
https://www.freevoiceusa.com/business_phone_systems/guides/admin/firewall_configuration/pfsense_firewall_freevoice_sip_phones
Please try the following to get your Freevoice SIP Phones working properly from behind a PFSense firewall. Disable source port rewriting - by default, PFSense rewrites the source port on all outbound traffic. Rewriting the source port of RTP can cause one way audio. In that case, you want to use manual outbound NAT and Static Port on all UDP ...
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