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sip - Asterisk keeps confusing public and private IPs so ...
https://stackoverflow.com/questions/61515693/asterisk-keeps-confusing-public-and-private-ips-so-calls-have-no-audio-when-soft#:~:text=The%20system%20works%20perfectly%20when%20set%20up%20on,open%2C%20no%20audio%20is%20heard%20on%20both%20endpoints.
Sip Client behind NAT (No audio until client unmutes ...
https://community.asterisk.org/t/sip-client-behind-nat-no-audio-until-client-unmutes/83751
When I call into the conference, my call gets connected but there is no audio because packets are sent to 172.18.0.6 address. ... The sip client is the one behind the NAT. ambiorixg12 April 24, 2020, 5:46am #4. creativeguitar: In my case is the other way around. The sip client is the one behind the NAT.
FreePBX behind NAT no audio at all - General Help ...
https://community.freepbx.org/t/freepbx-behind-nat-no-audio-at-all/68819
Hello, Everyone I am new to FreePBX and my box works fine with PSTN. Recently, I applied a sip.us trial account and tried to dial through sip but there is no audio and cdr show as answer. This box is behind NAT and all sip firewall rules applied, There is no sip connection blocked when I made sip call. Please give me some hint to solve this problem. Here is the log > …
SIP behind firewall no audio in both ways - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/sip-behind-firewall-no-audio-in-both-ways/td-p/4191197
SIP behind firewall no audio in both ways. Hello Everyone im new in voip im working in new project and i have this toplogy. Cisco jabber -->Cucm-->Cube-->Firewall (sophos)--->INTERNET (SIP-Trunk) the problem was the phone ringing but there's no sound in both ways and i think the problem is NAT. the firewall is sophos.
CUCM endpoints behind NAT (No Audio) - Cisco …
https://community.cisco.com/t5/ip-telephony-and-phones/cucm-endpoints-behind-nat-no-audio/td-p/2382830
Call established with No audio in both cases, enforcing trusted point relay had no effect. 2) Two Cisco IP communicators and Third party SIP phone (linksys spa 941) located behind a firewall (Nated to reach the internet) Control protocol is SIP: Registration fails.
Using SIP Devices behind NAT – The Smartvox …
https://kb.smartvox.co.uk/voip-sip/sip-devices-nat/
When working with SIP devices behind NAT, the ports that you may need to set forwarding for are: 1. The main SIP connection port – usually this is port 5060. The protocol is nearly always UDP. 2. The RTP media port or ports – often a …
Possible newbie issue - No audio on extensions behind NAT ...
https://community.freepbx.org/t/possible-newbie-issue-no-audio-on-extensions-behind-nat-server-tries-to-contact-private-ip/69403
Possible newbie issue - No audio on extensions behind NAT, server tries to contact private IP. I’m quite new to FreePBX and Asterisk so this could possible be a newbie issue solved easily, but I have searched the forums and manuals and not been able to find a solution to the issue. I’m running the FreePBX on a public IP (behind external firewall) and a few local …
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