Audio problems on BuyDID incoming calls (forwarded from DIDWW)

Pyheme

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Hi,
I'm struggling to get audio on calls to a DID originating from DIDWW and forwarded to IncrediblePBX 2021.
Currently, the calls are forwarded to a sip2sip address that get properly registered in freepbx and shows no audio issues when calling the 3333 and 4444 numbers.
I've also tried to forward the calls to other providers to no avail, including Callcentric which I otherwise use often.
The calls come in and get forwarded to the defined route but there is audio either way, be it IVR or an extension answering the call. After a few seconds, the call hangs up due to "retransmission timeout".

Attached are the asterisk details for a test call with an inbound route routing the call to extension 210 (pjsip). Can't paste them here due to length limit.

As instructed by BuyDIDNumbers, 2 SIP trunks were created with the following attributes:
USER Context: 46.19.210.14 (and 46.19.209.14)
USER Details:
Code:
type=peer
host=46.19.210.14
dtmfmode=rfc2833
context=from-trunk
insecure=very
nat=never
allow=all

Sip2sip provides some additional invite data where I could confirm that these IPs were from DIDWW.
I have call logs on sip2sip website for the same call but can't post in this message due to length.

In looking at DIDWW support pages, I noticed that they instructed to open UDP ports 10000-65535 (https://www.didww.com/Knowledgebase/ports_in_customer_ua_user_agents_or_firewall_post/) which I did and managed to have some audio in the very first call done immediately after, which got disconnected seconds later. I no longer could get audio after that one time.

I'm sure it's a NAT issue but can't find the proper settings to resolve it.
Today, I setup one of my Yealink phones in the same LAN as the IncrediblePBX server with the sip2sip account and calls come in without any problem...

I'm running out of ideas. This DID is intended mainly for inbound calls to our system. Forwarding it in BuyDID to another DID works but we lose the original CID which is less than ideal.

I welcome any ideas at this point to find a way to properly recieve incoming calls on this DID.
Thanks in advance!
 

Attachments

what is your router? do you have SIP ALG disabled?
 
what is your router? do you have SIP ALG disabled?
I thought about it too. Router is Asus.
Disabling SIP ALG helper blocks all SIP connection on my network.
Switching between "Enable" and "Enable + NAT Helper" does not change the issue.
 
I thought about it too. Router is Asus.
Disabling SIP ALG helper blocks all SIP connection on my network.
Switching between "Enable" and "Enable + NAT Helper" does not change the issue.
is your PBX server local network or in the cloud?
 
Are ypu sure you have the proper ports open on Iptables?
 
Are ypu sure you have the proper ports open on Iptables?
Important piece of information, you're right.
The server is local and behind a dynamic ip.
All commercial trunks I've set-up work as intended but this is the first time the DID is forwarded by a third party where I don't have a trunk to register to as such.
 
Between these this will be difficult to resolve without allowing anonymous SIP which is not recommended.
Thanks. I came to the same conclusion.
Today I setup a public-facing pbx on a spare dev box and confirmed that everything works there.
Interestingly, I could link both freepbx servers through IAX2 trunks and was able to recieve the call on the dev box and forward it to my server behind NAT without any audio issue.
It's far from being perfect but it does help to validate that NAT was the issue and, with that, I can guess with can close this topic.

Thanks all for your help.
 

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