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Just an idea, I have not tried this. Could you use a variable array with the returned gateway as the key? For example: (with reference to https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor) You call distributor with your list name... <action application="bridge"...

You have inspired me to go away and think about some creative regexp!

Hi James, I do use profiles on most customer accounts, maintaining all the keys etc. would be a nightmare otherwise, just didn't want to add any confusion here.

Just double check that in your external SIP profile you have set ext-rtp-ip and ext-sip-ip to your actual external IP address or set the equivalent variables. When you originally configured your external IP did you edit the values in Advanced->Variables?

I have always separated different tags with commas eg GW1, GW2, GW3. What this then actually does is set the ${toll_allow} variable to be "GW1: GW2: GW3". so then it's up to you how you use it. If you wanted the condition to be true if any one of the tags is present you might test like...

I thought that was the case but I had to ask the question to be sure. What I don't understand is why zero out DSCP on the public internet, why not just ignore it but leave it alone?

I think the reason 192.168.4.112 was in the Contact header is because your inbound call is hitting your internal profile on port 5060 not your external profile on port 5080.

Some of you may have noticed that the default FusionPBX installation includes some Iptables rules to set the DSCP byte to 46 (Expedited Forwarding) for outgoing RTP packets and 26 for outgoing SIP packets in the IPv4 header. Yealink phones also set these same QOS (Quality of Service) tags...

OK. I assume: the IP ending in 22 is your SIP trunk provider 192.168.4.112 is your FusionPBX 10.0.5.20 is your VoIP phone You see the 200 OK message from Fusion to the SIP provider is repeated several times, this is because Fusion is not seeing a response to the 200 OK from your SIP provider...

I can't answer for @TimGuyUK but here in the UK some SIP trunk providers will allow you to dial local numbers** without a city (STD) code. For example the Nottingham STD code is 0115 so you can, from anywhere in the country, call the Nottingham number 01159123456 but if you were within the...

If it helps, here is a (slightly altered) screenshot of the device settings for my desktop phone. There are four different extensions registered each on a different domain and there are some line keys programmed.

They are free text. Whatever you put in the Toll Allow field in the extension record or the Forwarding Toll Allow field in the ring group record just sets the ${toll_allow} channel variable. It is not implemented anywhere in the dial plan to it is entirely up to you what you do with it.

Yes, Wireshark is a very useful tool. I tend to use tcpdump to do the capture then use Wireshark or sngrep to look at the results. If I just want a quick and easy look at a few packets I use ngrep. sngrep will also capture but it has memory leaks and I find it will often segfault at just a...

Have you restarted the SIP profile? Or restart the whole machine to be sure, if it's not doing anything else. The best way to diagnose this is to look at a packet capture, sometimes referred to as a pcap trace. If you don't know how to do this, let me know, I will explain. It helps greatly...

Don't worry about the RTP port your provider uses. These are just the port numbers that the supplier will send RTP from. Fusion (FreeSwitch) will tell your provider what port to send RTP to in the SDP body of an INVITE or 200 OK message. If your Fusion is behind NAT, AND your SIP provider is...

Reading the Freeswitch Wiki it says "effective_caller_id_number is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_number."...

Ok, your ACK packet looks as if it is being sent to 116.20.30.42 port 5090, the IPv4 part of the capture that shows the actual source and destination is not shown, so for now lets assume that the packet is being sent to 116.20.30.42 port 5090. You have your external port 5090 forwarded to the...

There are many different ways you can do this, you will need to decide what best fits your needs. First of all, have a look at the documentation for Toll Allow: https://docs.fusionpbx.com/en/latest/additional_information/toll_allow.html I had a situation where I needed to stop just a few...

Nothing is leaping out at me as being obviously wrong with that 200 OK packet, but you say the ACK is still going missing... I assume you are taking your packet capture on the LAN side of your router. The next things I would try to do is get a packet capture on the WAN side of your router and...

This looks like a packet sent from the wrong SIP profile, you said the external5090 should be listening on port 5090. Also I notice in your firewall screenshot you have both external facing 5060 and 5090 forwarded to internal 5090. The SIP profile cannot put both in the contact header, it...