Your router is intercepting and changing the packets as they pass throughīe sure to Disable SIP ALG on the router.Use TLS encryption for the SIP signalling, especially when combined with a server running SIP on port 443.(this requires server side support for TLS).contact your internet provider and ask if they allow VoIP calls on their network.Your network provider is blocking VoIP calls. (this happens mostly in middle eastern countries).If you are not able to do so yourself, we would suggest contacting your ITSP (internet service provider) If not, you have to fix your internet connectivity issue(s) first. Please open your browser and check if you can visit a random website. There's an issue with your internet connectivity.But also I've also heard that VoIP can crap out on VPN, but I've not tried or experienced it.Open Zoiper -> Go to Settings -> Accounts -> (your account)ĭouble check that the setting for "Host"is correct and does not contain any spaces. I know I could create a simple solution by providing VPN access to my network, but I wanted to provide VoIP to my close family without granting them full access to my network. This is what is happening at the console (but from a different location and the external extension was 1000 this time): But I have no idea if the 3CX phones take advantage of UPnP. One thing I just thought about, I don't know if my brother has UPnP enabled on his router, mine is enabled. This could simplify my port forwarding and allow connections to multiple softphones behind my NAT. I don't even know if FreeSwitch acts or does not act as a proxy for the 'Voice stream' when the connection is from internal to external. I figured that if I could understand the networking behind VoIP, I could figure out a way to make it work. The voice from my network would reach the remote softphone behind the NAT, but I would not receive any voice stream from my brother's phone. I read that the voice is trying to go peer-to-peer using RTP so then using router configs we forwarded the RTP ports (40000-40049) to the PCs with the soft phones on both ends. The SIP signaling worked fine, the phones were connecting, authenticating and ringing, but no voice. In the configuration used, this should make his phone part of "My PBX". I had the remote 3CX phone configured to be a client to my Freeswitch using extension 1019. With a vanilla installation of FreeSwitch, I was able to get the 3CX soft phones to work inside my network but I had mixed results when the 3CX soft phone was behind another NAT router (my brother's). I'm trying to use my Linux-FreeSwitch box behind my home NAT router and it does not work (yet!). I hate the interface, but it gets across the NATs. So if you want to stick with a standalone FreeSWITCH install, I suggest to try using the X-Lite softphone. it's NOT designed to run on YOUR linux installation.Īlso note that I was not able to get SipXecs to work with the 3CX client outside the LAN either, but the X-Lite client from CounterPath does work. It might not please everyone, as this solution creates a dedicated PBX computer, i.e. I have elected to resolve the problem with SipXecs It includes CentOS + FreeSwitch + NAT traversal plugins. This article was created by MediaWiki user "Politick" on 25 February, 2011 FreeSwitch, Networking, and NAT
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