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The port should not matter at all, if that has fixed it it is pure luck or a setting in the other profile.

If the kamailio is the .4, kamailio is auth challenging the freeswitch, First thing I would do is switch that zoiper transport to TCP you are getting frag on the UDP

My guess is that Kamailio is trying to reinvite for some reason, can you get a siptrace of that call from sngrep. The kamailio is using the wrong domain though for the digest auth by the looks of things.

That is your problem ^^^^^^ No idea what is going on there. Is there some sort of forwarding going on or something?

Yes, its before that that I want to see, from the beginning of the call preferably.

Look in: /etc/cron.daily/fusionpbx-maintenance

iptables?

Call forwarding/follow me, when set on an extension, only works with the enterprise ring group strategy.

@mrjoli021 Can you post a fuller log of an inbound call, also set the zoiper transport to TCP. Zoiper actually seems to work the best from all my testing. Zoipers push servers are 185.117.83.192/27 so what you are seeing is perfectly normal.

You can do this

There are lots of things to consider, you will just have to monitor it and watched pdd etc. For example, there is a huge difference in a remote datacenter that is 15ms away vs one that is 150ms.

Its very likely a NAT issue and just switching to TCP as the transport on your phones will probably fix it immediately. By the sounds of it your NAT tunnels are timing out on the router. By restarting the sip profile etc, you are just refreshing those tunnels. Many devices have very low...

I would maybe do it something like this... When a call is made, store the extension number in redis with an expiring key for the length you need, eg last hour, day week etc. Then when new call comes in, look into redis for that callerid and find the extension that matched it, then route to...

I also wouldn't mind betting asterisk is far more resourced than Freeswitch too. Also, much of the failings of asterisk, from my limited understanding, revolved around the chan_sip module. Now that its using pjsip, things may be much different.

My interest is stability, I think it is now well over 3 years since we have had a stable release of FusionPBX. I want a PBX that I am not terrified of upgrading. My only concern at the moment re vitalpbx is cross datacenter availability but I'm sure I will be able to skin that cat.

I have installed this, it threw no errors and seems very slick and IS multitenant. I'm very impressed with it so far.

Have any of you guys tried call transfers yet? Many of our calls go to a different person initially and this is where most of our problems lie currently. The calls, in general, first go to a ring group where someone picks the call up, then transfers it to the desired endpoint. The transfers can...

You have posted no logs or opensips config. The person who can fix it for you with that limited info is a genius, that is why you probably have no replies.

You don't register to Simwood usually. In the simwood portal, set the did destination to %e164@yourdomain.com:5080, and disable opus too unless you really, really need it. It adds 239 bytes to the packet. Make sure you add all simwoods ips to your ACL which you can find in their knowledgebase.