VoIP call Continuity and fail-over to second Internet Connection

Hi Everyone ,

This is a non related Freepbx system but a SIP/RTP Question.

Is there a away to to keep an active call alive without dropping it, if the internet connection it uses fails and then route it to another internet connection? Is this something that is supported in SIP in general ?


There is no way asterisk can do that, it is a b2bua so needs to be in the path all the time, a true proxy would hand off the ‘session’ between the caller and the callee.

Thanks Dicko for your reponse.

Can you advise any SIP proxies that could support such?


My preference is Kamailio.

Not True. I don’t know how it’s working but I have accomplished just that.

As a test site I have a ubiquity router edgerouter PoE and configured Eth0 for my main connection and Eth1 as my failover. Eth0 is with Verizon fios eth1 is with spectrum. Eth2 is my LAN where my sangoma s700 sits. So I disconnected the primary wan while on a call and I lost about 1 to 2 seconds of audio before the audio resumed. I will do a tcpdump and find out how or why this works.

Now this will only work for remote phones. Next step is to test for the pbx to work the same way.

How’s that going for you so far?

It’s working still. It works perfectly with Sangoma And Polycom phones. Haven’t had any luck with yealink or Grandstream phones.