Remote VoIP Phone Unifi Network
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@NashBrydges said in Remote VoIP Phone Unifi Network:
I've port forwarded 5060 to the FreePBX server limiting access to the remote public IP
You have to port forward the RTP range also. It is UDP 10000-20000 by default.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
Asterisk info shows the remote extension is registered however "UNREACHABLE" (see screenshot)
This is not registered. this is an issue with NAT.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
I've added the remote IP to the "networks" tab on FreePBX
Have you restarted the server or issued
fwconsole restart? -
@JaredBusch Yeah that was the first thing I tried but didn't help. I am going to port forward the RTP ports on the FreePBX network and give that another try.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
FreePBX v14.0.13.40 setup and running onsite with 55 extensions.
FYI, this needs updated.
https://www.mangolassi.it/topic/21589/upgrade-freepbx-14-to-freepbx-15 -
@NashBrydges said in Remote VoIP Phone Unifi Network:
I am going to port forward the RTP ports on the FreePBX network and give that another try.
That should resolve the RTP audio issues.
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@JaredBusch Are you suggesting that an upgrade could fix this issue?
I port forwarded 10000-20000 to the FreePBX and still no joy.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
Are you suggesting that an upgrade could fix this issue?
No, the things are unrelated. Simply behind by almost a year is not normally a good thing.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
@JaredBusch Are you suggesting that an upgrade could fix this issue?
I port forwarded 10000-20000 to the FreePBX and still no joy.
Then you have something wrong in your network settings, because no RTP is 100% a NAT issue.
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@JaredBusch Yeah, I'm relatively new with this client and trying to fix their most critical problems first. Agree, that needs to be upgraded.
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Have you considered putting a VPN between the sites? That will remove the NAT completely.
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@scottalanmiller said in Remote VoIP Phone Unifi Network:
Have you considered putting a VPN between the sites? That will remove the NAT completely.
@scottalanmiller - Agreed, especially if you have unifi at both sites it should be quick and easy (assuming they are on the same controller).
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@NashBrydges - What phone are you using? Is it a Grandstream by chance?
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@syko24 said in Remote VoIP Phone Unifi Network:
@scottalanmiller said in Remote VoIP Phone Unifi Network:
Have you considered putting a VPN between the sites? That will remove the NAT completely.
@scottalanmiller - Agreed, especially if you have unifi at both sites it should be quick and easy (assuming they are on the same controller).
Exactly, that's what I was thinking.
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@scottalanmiller Yes I have, and given the phone is a Yealink, I think that capability exists however this is not my ideal workaround. I'd like to fix the actual problem without having to go VPN. But if there is no fix that I can find, I may not have any other choice.
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@syko24 Yealink T57W
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@NashBrydges said in Remote VoIP Phone Unifi Network:
I've added the remote IP to the "networks" tab on FreePBX
My brain just caught up.
Remove this. It is not a "local" network because it is going through NAT.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
@scottalanmiller Yes I have, and given the phone is a Yealink, I think that capability exists however this is not my ideal workaround. I'd like to fix the actual problem without having to go VPN. But if there is no fix that I can find, I may not have any other choice.
I shoudl clarify that I don't want a site to site VPN. The only acceptable VPN solution would be the device VPN.
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@NashBrydges said in Remote VoIP Phone Unifi Network:
@scottalanmiller Yes I have, and given the phone is a Yealink, I think that capability exists however this is not my ideal workaround. I'd like to fix the actual problem without having to go VPN. But if there is no fix that I can find, I may not have any other choice.
The phones have nothing to do with a site to site VPN between two routers.
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Also go into the Asterisk SIP Settings and ensure that NAT is enabled for
chan_sipif you are using it.
