Client side router - SIP Failover

[gflow]

Hi,

I am testing fail-over to 4G backup internet from my primary fibre service. The fail-over works fine with my router however I have noted the following and wanted to know if there are any settings within FusionPBX I can change so that my calls aren't dropped during a call.

* If I place the call on hold and then fail-over to the backup 4G service, it works perfectly no lost audio.
* If im on a call and fail-over there is no audio at all unless I place the caller on hold and then resume, I can then continue the call.

I'm hoping there might be a few SIP experts on this forum who can assist here.

Note: Public IP Address is changing during failover.

 

[EasyBB]

I believe hold-resume will initiate a RE-INVITE

 

[DigitalDaz]

Not that I know of, this is a NAT mapping issue. You may get away with pressing hold on and off, that may generate the reinvite but other than that I can't see anything. How could the PBX reinstate a call to an IP, (your new 4g one) that it isn't even aware of at this point.

 

[gflow]

Not that I know of, this is a NAT mapping issue. You may get away with pressing hold on and off, that may generate the reinvite but other than that I can't see anything. How could the PBX reinstate a call to an IP, (your new 4g one) that it isn't even aware of at this point.

That's what I thought as well but if I have a call placed on hold and then disconnect the primary link the music on hold does not skip a beat and then I take the call off hold and im back talking.

I also read something on an Asterisk forum where they were suggesting to somebody in a similar scenario that RTP packets hold the source address information and calls shouldn't drop and the PBX will know the new IP address due to information contained in the RTP packets.

 

[DigitalDaz]

I reallt wouldn't worry about it. Most people will just look at the phone, curse it and redial