Newbie here. i’m using FreePBX 2.11.0.11, Asterisk 11.
We are having intermittent audio issues, at times one way audio or even no audio at all. we are experiencing this even within extensions calling. Our server is located outside of our network and other extensions are on different network as well. we observed this problem especially when our pbx server losses brief internet connections. But it’s also happening on a random basis even if the server doesn’t loose internet connections.
We are also having one way or no audio issue on Cisco SPA525g2 5 lines SIP phone(located in a different network). the phone is configured with 3 different user extensions (lines 1,2, and 3).
line 1 seems to be working fine, however, lines 2 and 3 are having one way or no audio at times and would need to perform factory reset, then reconfigure the lines 2 and 3 again for it to work. And we’re experiencing this issue over and over.
Sorry, I’m pretty new to this stuff, not sure if i understand you correctly or not. As for setting up VPN, are we going to setup VPN on the FreePBX server itself or on the network where the SPA525g2 is located?
To give you a clear view of our network. Our FreePBX server is hosted by HE, a group of extensions are located in a remote network, and the Cisco SPA525g2 is located in a different remote network.
maybe you could provide me a documentation or link on how to setup a VPN.
Thank you for having interest in helping me, appreciate it.
Not sure if they offer IPsec connections, will have to ask our network admin.
If i may ask, what is IPsec connections and what is that for?
Sorry for my ignorance about IPsec connections. If you could please give me a little explanation about IPsec connections, and how does it affect audio issues on voip communications.
but if your network person can handle session timers and opening ports you should be able to get this to work as well. however the security issues are more complex if you are not using vpns. the only negative of a vpn is that it does require more network bandwidth
aside from vpn, you’ve mentioned the use of session timers.
i am not familiar with session timers.
if i may ask, how can we effectively setup session timers so that we could eliminate one way or no audio issues? Please explain it a little further…
to close ports or sessions when they are not in use. of course this discussion depends on the type of firewall you have in use. when the phone registers with the pbx it “opens” a udp port, typically port 5060. this starts the session timer in the firewall. if the timer expires the firewall closes the session. now if the pbx tries to send a call to the phone the port is closed and the phone will not ring. you want to set the udp session timer on the outbound (lan->wan) rule to be at least the length of time between registrations initiated by the phone. a rule of thumb is to set the timer at 300 seconds.
Thank you very much guys for helping me out, really appreciate it.
If i may ask, what would be the best way to eliminate intermittent audio issues(one way or no audio) without having to use VPN and session timers in the firewall?
By the way, we are just using a handfull of extensions (less than 20), and the SIP phones we’re using are Cisco’s SPA303, SPA232D, SPA504, and SPA525g2.