I realize this question might be more appropriate in a Ubiquiti Forum, however I thought I’d post something here in case someone has had a similar experience.
I’m looking for help is setting up my EdgeRouter Lite to work with FreePBX. Specifically what configuration on the Edge Router Lite will allow username & password authenticated trunk to work / register properly with my SIP provider. I don’t think I require port forwarding however do I need to put the FreePBX into a DMZ and if so how and I prefer to use the GUI to do this? I do have “eth2” available on the router if that helps. Also does anyone know what role the FreePBX firewall will play in the set up? Currently my SIP trunk is showing as registered with my provider, however at their end they keep seeing it register then un-register. Recently I defaulted the Edge Router and used the “Wan2Lan” setup wizard to configure the device.
Any information or links to video’s / documentation would be greatly appreciated. Thanks in advance for your help.
Update your router firmware to current stable 1.10.8
Default your router again and use the basic setup wizard, it is the current version of the custom ones below it.
Those other configs were community created a long time ago. The Basic Setup Wizard was added in umm 1.9.something.
That’s it.
Nothing else needs done.
If, and I mean, if you have an issue with your trunk, you can disable their SIP ALG helper. This is one of the only routers that I never have any problems with.
Open SSH to your router
configure
set system conntrack modules sip disable
commit;save;exit
You do not need DMZ, or port forwarding, or anything else withmost SIP trunk providers that use registration. VoIP.ms, Twilio, Flowroute (i believe, been a while) are ones I have used.
Unless you have a super compelling reason, you should always turn of SIP-ALG for a server configuration. These helpers are usually a little too helpful, and they cause early baldness.
I do agree with this, and I have it turned off in all EdgeMax devices I have control over. But I have come across multiple sites with these in place and no issues without it being disabled.
certain type of phones love SIP ALG… the cheap ones…and depending on the firmware of some good ones. With my experience Sangoma/Yealink/Polycom phones hate SIP ALG
Just want to clarify something. We’re talking about FreePBX and Asterisk. SIP-ALG causes all sorts of weird, unreproducible errors for just about any phone that connects through a router that uses SIP-ALG to Asterisk.
Now, I don’t disagree that SIP-ALG can be useful for some inexpensive phones with other SIP providers, but for us, it never adds value, and while it doesn’t hurt you occasionally, it will eventually screw up a call when working with Asterisk.
So, unless you have a compelling reason to use SIP-ALG, it should almost always be turned off when working with Asterisk.
Hello
I’ve turned off SIP-ALG. I can see my SIP trunk registered with my provider however at his end he see it as registered and then not registered. Whenever I place a call, I get “All Channels Are Busy” message.
@sorvani , Excellent!!, now all those who have had more than “zero issues” with them, all anticipate your actual solution as to “how to do that 100% thingy”
Why are you not using a chan_pjsip based trunk? Have you tried that?
If you are going to stick on the chan_sip trunk:
Remove permit and deny.
Don’t overcomplicate things when things are not working. Also, the FreepBX firewall should handle restricting most connections.
You are behind NAT, assumption because you have a router you are complaining about, but you did not specify that the trunk should use nat.
Hello
I’ve got outbound working successfully however inbound isn’t.
As a back up, I can connect to my provider with an Cisco SPA 122. When I look at the log file I see the 10 digit CID and the FreePBX routes successfully on this number.
If I disable the Dahdi trunk and enable the SIP trunk, then the 10 digit CID isn’t found in the log file and the caller gets a rejection message. Is there something in the Sip peer details that could be preventing this?
Hello
Just a quick update. I haven’t been able to get inbound routing working when I connect up my SIP trunk. Inbound calls work fine when I’m connected using an ATA, however when I move the trunk off the ATA and connect it directly to the FreePBX, inbound routing doesn’t work. So there’s a difference in the setup between the ATA and FreePBX when connecting a Chan_SIP trunk. Is it my FreePBX configuration?
Thanks in advance for any help that can be provided.
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