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No audio on sip calls over VPN - Endpoints - FreePBX ...
https://community.freepbx.org/t/no-audio-on-sip-calls-over-vpn/78414
No audio on sip calls over VPN. i have a FreePBX (asterisk) system as my pbx. It is connected to my Mikrotik. I have two Mikrotik i have setup server l2tp VPN and client VPN. Inside my internal lan, 10.0.0.0/24, everything is working fine as voip telephony concerned. When i connected through VPN, i can register my sip phone and i can call every ...
SIP over VPN - One Way Audio (Incoming only) - Asterisk ...
https://community.asterisk.org/t/sip-over-vpn-one-way-audio-incoming-only/82022
I’m new to Asterisk but have come a long way in a short time. I’m using Asterisk 13.29.2. I’ve set it up and have it working perfectly with standard internet connection. I am now trying to place calls from the same PC (X-Lite) through my VPN and only have one way audio (incoming). I have added the following lines to sip.conf trying to get it working, but the problem …
Asterisk through a VPN is not working - Asterisk SIP ...
https://community.asterisk.org/t/asterisk-through-a-vpn-is-not-working/89469
If it is no audio, you should turn on RTP debugging and confirm that packets are being sent and received on both sides. That should help show where they are getting lost. The pjsip set logger on output is also going to be helpful, in showing which addresses are being sent.
One-way Audio over Site-to-Site VPN at Remote Sites ...
https://community.freepbx.org/t/one-way-audio-over-site-to-site-vpn-at-remote-sites/51675
I don’t believe it to be a router or VPN issue because I’ve been able to take a phone at my Main Site, register it to Asterisk 1.8.28-cert2 at a Remote Site over the VPN, and make calls with two-way audio to everywhere except the phones at the Remote Sites that are registered to FreePBX at my Main Site.
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not hear one each other, I used from-internal context for both of them.
No Audio for SIP calls through OpenVPN - OpenVPN Support Forum
https://forums.openvpn.net/viewtopic.php?t=13367
3) Unembedded freePBX -> login -> Tools menu -> Asterisk SIP Settings 4) Select autoconfigure to populate external IP and local datacenter network. 5) Click add local network field (in my case non-datacenter LAN-LAN VPN networks) 6) Add 192.168.8.0 / 255.255.255.0 and 10.243.250.0 / 255.255.255.0 7) Save and apply changes
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