Thus you would expect to see rtp traffic between the To: and the From: on the port in the m= line, 192.168.2.35 to 192.168.2.35 on 16400 is probably not going to work but 192.168.2.1 to 192.168.2.35 on port 12954 is probably what you are seeing
Double check your extensions, your router NAT rules and your channel configuration
Thank you for responding, but Iâm not sure where to look exactly. My router is allowing UDP traffic on ports 10000-20000 to my PBX address and it shouldnât block any outgoing traffic. I noticed I switched direct media back on in the last test, but here are two new examples of direct media set to no on both extensions. Iâm noticing m=audio 60124 in the log and this port should not be covered by the allowed range, where is this coming from and what do I need to check exactly?
After quite some experimenting, I now have somewhat of a weird setup that seems to work fair.
NAT SIP settings in FreeBPX:
External Address: 188.195.237.73 (no idea what IP this is, but only with this configured I can receive inbound calls)
Local Networks: 192.168.2.0/24 and MY_PUBLIC_IPV4/32 (this works since the DECT/router extension is calling âfromâ my public IP)
There is only one problem with this configuration. Inbound calls disconnect after 32 seconds or when mic is muted for a few seconds. Outbound calls and calls between extensions work flawless now as far as I can tell.
Do you have a clue?
EDIT: Right now the audio stops after even 10 seconds.
Those are signs that your routerâs NAT forwarding rules are not correct.
My router is not blocking any outbound traffic and is blocking all incoming traffic except for ports 10000-20000 UDP to my PBX. I am not allowing port 5060 at the moment due to local-only environment and security risks, is this necessary? What am I forgetting here?
Rules for keeping connections open through routers performing NAT can be of many flavors, for example the SDP/RTP traffic might be considered âAssociatedâ with the SIP session and not need any specific forward rules (Deep Packet Inspection), some routers have socalled âhelpersâ that incorrectly route VOIP calls , keep alive packets need to be honored and so on, you should look for a router specific recipe.
I am using a ISP provided modem/router which works great. SIP ALG is disabled on this router. There are no further settings related to NAT. Shouldnât the current configuration already be enough?
Apparently not 
Haha yeah sure! I donât have a clue on what to change/check anymore. I think it may have something to do with my external ip being used by the DECT extension locally as well. As stated, currently I have 188.195.237.73 configured in the PBX as my external IP, which is not really the case, I have no clue where this IP comes from but it was the default and only with this IP the inbound calls come through (but drops after 33sec).
I donât think this is a router issue but rather a misconfiguration in the FreePBX SIP settings. Can you follow me?
whois -h whois.cymru.com â -v 188.195.237.73â
AS | IP | BGP Prefix | CC | Registry | Allocated | AS Name 3209 | 188.195.237.73 | 188.192.0.0/14 | DE | ripencc | 2009-06-08 | VODANET International IP-Backbone of Vodafone, DE
Not really, can you find a peer using the same provider, pressumably Vodaphone?
What do you mean by that? I donât have Vodafone as provider.
You definitely mentioned
I just looked it up for you.
Yes, I did mention this IP address, but this is not my IP and Iâm not related to Vodafone either. What should my next steps be? Happy to hear your ideas.
Dicko, do you have any clue whatâs causing this issue and how to resolve it? Happy to hear back from you.
Not really if they are indeed local but it seems that your âlocalâ extensions are not necessarily all technically local (directly on the LAN) so would look at the intervening routers.
Okay, so now trying to use the systems with softphones only to rule out the router/dect extension calling from the wan ip. I notice that I only receive inbound calls when I set this unknown IP 188.195.237.73 an my external address in the sip settings. When this IP is set up the inbound call terminates after 30/32 seconds, outbound calls work fine.
I think because this IP is not my actual WAN IP, the value in the contact header is wrong and the call terminates because of that. But, when I set up my actual IP as external address there is no record in the asterisk logs and I get the message that the number is currently unavailable.
So, I think I need to find out why only this strange IP allows inbound calls and not my actual IP. Do you have any idea? I am using RasPBX as distro.
Possibly, your ISP is doing CGNAT and/or caching.
If your external IP address is between 100.64.0.0 and 100.127.255.255, thatâs CGNAT.
If your external IP address is between 10.0.0.0 and 10.255.255.255, thatâs a private IP address.
If your external IP address is between 172.16.0.0 and 172.31.255.255, thatâs a private IP address.
Report any address thatâs not your external IP address when visiting
http://checkip.dyndns.org/
http://checkip.dyndns.org:8245/
https://www.whatismyip.com/
Also, it would be useful to see the Asterisk log of a failing incoming call, with pjsip logger turned on. I tried to look at your previous pastes, but they somehow expired, even though the default expiration on pastebin.com is Never. Possibly, youâre a member and have an expiration set for your account. In any case, after pasting the log and before clicking Create New Paste, scroll down to where it shows Paste Expiration and make sure that it is set to Never. Among other things, this will enable future readers of the thread to follow along.
Hi there! I donât have CGNAT. Just tried the system without calling from WAN and have the NAT settings as itâs supposed to be. It now seems to function great. Thank you for any assistance.
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