Ok, I have 2 soft-phones both are Media5-Fone on the iPod touch. This is set up so that wireless phone calls can be made over WiFi. I have a bind port set up as 5061 in Asterisk.
The 1st one is extension 4000. It is set up in the extensions as such and the iPod touch connects just fine, even remotely. I have that port forwarded on the router and firewall.
The 2nd is extension 4003. It is set up exactly the same as the 1st but with extension 4003 instead of 4000.
If I call 4000 from any phone on the PBX, it goes to voice mail right away. If I set both phones up in a ring group, only the 2nd one rings.
The funny thing is before I had both the 1st and 2nd soft-phones, I was using just the first soft-phone and it was flawless.
FreePBX has nothing to do with your network directly, asterisk (SIP) will need a unique IP adrdress and port pair for each extension to function (register) correctly, you say “. . THE iPod . .” if it is one iPod then Asterisk (sip) will send it to that device, If I am correctly reading what you state, use a different port for each “softphone” that seemingly is behind the same IP address.
When I said THE ipod touch, I meant each softphone is an ipod touch (confusing, I know. Sorry).
What is the difference between the bind port setting and the port setting in the extension.
Here is what I have:
Asterisk - SIP Settings - Bind Port: 5061
Extension 4000 - Port: 5061
Extension 4003 - Port 5062
Router - Port Forward Range - 5061-5062 to
EDIT: Even though I used the same extension port setting as the bind port setting with one phone and it worked, with two, perhaps bind port should be 5061, extension 4000 should be 5062 and extension 4003 should be 5063. I am unsure.
I thought i mentioned the routing table above. If you need more info, I’d be happy to provide it.
Externally (WAN, say from my friend’s house who has wifi) i want to connect and make phone calls. Assuming my internal and external ports are the same (some routers allow different port numbers to trigger other ports) I should be able to connect using the WAN IP address of the PBX using the assigned port.
My questions are:
Is the port i connect with using the softphone the extension port or the bind port?
Do i have to forward the bind port? Further, do i have to forward any other ports.
Asterisk will listen anonymously to your bind ports, it will also honor requests outside to endpoints it knows about, the media stream will be negotiated and used post registration and initiated by that SIP transaction, in a “standard” asterisk implementation rtp on a port between 10000 and 200000 , it is your responsibilty to ensure that these processess will be transparently translated through your PNAT enabled router, each and everyone is more or less capable of doing that There are many routers that are incompetant in that area, that will unfortunately be your problem.
You should get it working without touching the bind port and certainly not the extension port settings.
If you want to change the SIP port Asterisk listens on then change the bind port and leave everything else alone. You have no reason to change the response port in the extension as each IP NAT’d will have a unique translaton.
Think about it this way. If you had a web server and 20 people connected to it, and the web server was behind a NAT. Would you have to use 80 different ports? Of course not.
You are creating your own issues by changing ports.
