WAN Drops, Keepalive

[ChrisLab]

Hi All,

I'm sitting with an issue where the ISP drops the WAN connection on router for a split second, and then reconnects again. They do this once a day. When this happens, the client's external IP stays the same.

After this happens, the phones drop connection to fusionpbx, and don't come online again.

If you log into the phone is shows Registered, But on Fusion its not registered.

I've got ping enabled under extension settings, but it seems the only way to keep the phones connected is to turn on KeepAlive(OPTIONS) on the phone.

I would like not do this for every client to keep SIP msgs at a minimum.
Feels like im doing two wrongs to make a right.

Anybody else experienced this, or has some advise?


Thanks!!

 

[Adrian Fretwell]

I have never used the ping option on the extension record, but I always use the KeepAlive(OPTIONS) on the phone. The OPTIONS/NOTIFY ping is only ~500 bytes in each direction every 30 seconds, it's not a big deal.

 

[ChrisLab]

Hi Adrian,

Thank you for your input.
I previously connected my clients to an Asterisk system, using basically just username, password, server, and everything worked out of the box. I'm finding with Fusionpbx, there's a lot of tweaking that needs to be done on the phone side, ex. Keepalive options, rport. Also had the issue today where, if you pull a call with *8, you don't get the caller id of the original incoming call. Had to change settings on the phone to get that working. If this is standard practice to do a bunch of extra config on the phone side, that fine, I just want to be sure that I'm not making mistakes on server side.

 

[Adrian Fretwell]

Hi Chris,
I understand your view. I have not played with Asterisk for over twenty years, so I struggle to remember what ti was like. I did like the IAX phone protocol, it just seemed to work and did not get broken by NAT in the same way that SIP does. But the whole world went SIP shaped and we have to put up with it.

I always found it helped to think more about FreeSWITCH itself than the GUI wrap around it. There is often more than one way of dong something and the method chosen by any given GUI wrap may not necessarily be the best option for your use case. Group Intercept is scripted in some setups, but it is directly available in FreeSWITCH:
https://developer.signalwire.com/fr...lained/Dialplan/Call-Group-intercept_7144635/

 

[ChrisLab]

Hi Adrian,

Thanks for the info provided.

I'll definitely look at getting to know the underlaying Freeswitch better.
At the moment I'm just trying to familiarize myself with how everything works, or should work.

Thanks again for your help, and for pointing me in the right direction.

Cheers!