[AX86U] Half audio on incoming VoIP SIP calls, NAT mismatch?

[RMerlin]

I was hoping to minimise my monthly outgoings by just paying for the SIP trunk and purchasing my own PBX as I am an unpaid carer. Renting a cloud PBX does add a little extra to the monthly cost.

Not really. My phone service (including the PBX features) costs me around $6-$8 CAD a month (I can`t give you an exact amount since I fill my account with $75, which get used over the course of 6-10 months). That includes the DID, multiple lines, the IVR to handle my two extensions, etc...

 

[DanielCoffey]

A shame it isn't that cheap over here in the UK. Mind you we have the 20% tax applied to everything first.

 

[RMerlin]

A shame it isn't that cheap over here in the UK. Mind you we have the 20% tax applied to everything first.

They are also available in the UK, although they have fewer POPs available in Europe. I don`t remember if they have a POP in UK specifically They have a POP in London. Since they have only a single POP I'm not sure how the performance will be for you, you'd have to try them out.

https://voip.ms/

 

[capncybo]

UPDATE : Sorted... almost.

GoTrunk were able to assist me with the "Unauthorized" error. In one of the older Guides I had followed it had advised me to turn on "Auth Trunk" in the Trunk Settings... "If enabled the UCM will send a 401 response to the incoming call to authenticate the Trunk". Leaving this at the default Disabled sorted the issue.

Now I was getting a "User Busy" error on incoming calls. This was being caused by my Inbound Route not handling the incoming Caller ID properly and no rule catching it.

If I had an Inbound Route of "_X." (any number of digits) it was failing because the Caller ID was being supplied as +441234... which of course does not match that number.

Now I just need to know how to handle the inbound Caller ID so that 01234... gets passed to the handset.

Sorry to interrupt your pleasant VOIP/PBX chat with RMerlin but...
Previously... I did suggest you (could/should) try the PBX by itself (on-its-own).
The intention was to verify if the PBX works independently & rule out any asus caused firewall issues.
IMO it's very probable your issues have little or nothing to do with your: AX86U
Hence (from a forum perspective) you could be... barking-up-the-wrong-tree.
But... you never bothered to answer any of my queries ???
Helping someone often requires a 2-way effort.
Don't get me wrong.... I sincerely hope someone with BOTH Merlin firmware interests & GrandStream PBX experience will see your S.O.S.
But it's rather unlikely they'll show up & save the day.
I certainly wouldn't wait around here hoping for the solution(s).

 

[DanielCoffey]

I do appreciate the suggestions and you were correct - the router was doing its job properly once I got the SIP ALG settings correct.

I am happy to report that GoTrunk (Trunk Provider) were able to help with understanding the pcap traces, folks here gave me enough guidance that I could understand what the router was being asked to do and now I have got the PBX itself behaving as intended.

The issues were mainly caused by both my Outbound and Inbound routes not catching the calls due to the fact that GoTrunk strictly apply the +44 rule for UK numbers. Now all I have left to work out is whether I am still aflicted by the one way audio through the NAT but my brain is too frazzled to deal with that today.

 

[roundaway]

Just a thought. There's a Germany and Netherlands data center. https://nerdvittles.com/the-5-minute-pbx-in-the-cloud-platform-for-2-a-month/

 

[Damit1]

I'd recommend Zadarma.
Works for me like a charm.
Great prices (maybe the best I've encountered?), great service as well.
Trully recomended

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[DanielCoffey]

Thank you folks. I am already set up woth GoTrunk for £5.99 inc VAT and have the PBX, handset and intercom (we have a disabled occupant). The copper landline is gone now as an added bonus.