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Pfsense and Asterisk….one-way audio problem with SIP ...
https://forum.netgate.com/topic/465/pfsense-and-asterisk-one-way-audio-problem-with-sip
Pfsense and Asterisk….one-way audio problem with SIP. This topic has been deleted. Only users with topic management privileges can see it. I have a setup with an Asterisk server in the DMZ. This Asterisk server connects to 3 SIP voip providers…Sipgate, Gossiptel and Broadvoice. Unfortunately I have a one way audio problem with inbound calls....ie I can't hear …
pfSense port settings for Asterisk FreePBX - Outside Open
https://www.outsideopen.com/pfsense-asterisk/
If you have audio only in one direction, take a look at the RTP port settings shown below. Option A: pfSense in an environment where you have multiple public IPs and with one IP assigned to your Asterisk / FreePBX or Avaya system. Option B: Port forwarding on pfSense for single IP system like you would have on a home Internet connection.
One-way audio behind pfsense firewall | Page 2 | The VoIP ...
https://www.voip-info.org/forum/threads/one-way-audio-behind-pfsense-firewall.10304/page-2
This is a long-standing disagreement between me and the pfsense devs, which I have given up on. They insist it be this way and that there is almost never a valid reason to use static port. Well, a lot of people have voip providers that cause one-way audio if you don't use static mode, and the argument about TCP stacks not randomizing ports well ...
Asterisk behind pfsense (no sound) | Netgate Forum
https://forum.netgate.com/topic/37420/asterisk-behind-pfsense-no-sound
I have 1 asterisk server behind pfsense nat and also 2 sip phones behind the same nat. asterisk server is at 192.168.20.248 and listens on UDP 5060 and RTP is 17000-18000. I am having a hard time getting this setup working – lots of SIP trunk registration timeouts, or no-audio problems when answering incoming calls.
One-way Audio SIP - OPNsense Forum
https://forum.opnsense.org/index.php?topic=304.0
My topology is this: CLOUD -> Switch -> PBX (Asterisk) '-> OPNSense -> Sip Client. When making calls I can not hear the audio from the PBX. This is what I tried: - Switch to Manual Outbound NAT rule generation in NAT Firewall. - Create a rule for static port. Note: As similar to pfSense I have tried to follow the steps indicated on this page: …
One Way Audio, No audio, Multiple DIDs, Dropped Calls ...
https://community.freepbx.org/t/one-way-audio-no-audio-multiple-dids-dropped-calls-after-30s-busy-on-incoming-inconsistent-results/65146
One Way Audio, No audio, Multiple DIDs, Dropped Calls after 30s … The VLAN has wide open rules to the rest of the networks. Basically, these four problems are all common and usually repairable relatively simply, once you know where to look. In fact, they can all be caused by an incorrect NAT setting.
One way SIP or dropped SIP after 30 or so seconds : …
https://www.reddit.com/r/PFSENSE/comments/6s6rgg/one_way_sip_or_dropped_sip_after_30_or_so_seconds/
One way SIP or dropped SIP after 30 or so seconds. Hey guys, I've installed a new set of pfSense (v2.3.4) firewalls with an IPSEC VPN between. SIP calls seem to work for about 30 seconds before call drops. Or caller can't hear our end of the call. All of the normal goto settings for VOIP don't seem to help.
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.
Yealink T23G openvpn pfsense freepbx one way audio ...
http://forum.yealink.com/forum/printthread.php?tid=41037
RE: Yealink T23G openvpn pfsense freepbx one way audio - jolouis - 08-04-2017 03:57 PM Sounds like your setup is correct for the VPN, but it might be a firewall or Asterisk setting that is causing the issue. If the phone registers and you can see it online, then I expect it is not a problem with the firewall.
One way audio, can't figure it out - General Help ...
https://community.freepbx.org/t/one-way-audio-cant-figure-it-out/12934
I have a one-way audio issue that I can’t figure out. Also the calls automatically drop in ~ 30 seconds. Calls from “outside” can hear me, but I can’t hear them. If I let a call from outside roll to voicemail by not answering, they can leave a message and there is audio in that message. Here is my rtp.conf [root@freepbx asterisk]# cat ...
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