im having an issue where a I have a remote connection for a SIP extension behind a router but there is sometimes intermitten connectivity from that remote site (its out on a gsm router and sometimes the signal is bad).
The extensions registers fine and inbound and outbound calls work fine as well, it just when ever there is a major drop on the routers end it , fails to reregister properly , by this i mean a sip show peers show unrechable .
Strangely enough i am able to call out from that number but when ever i try call in it thinks the extension is off .
I would be very appreciative if someone could point me in the right direction of how best to begin fixing this as i have no idea what would cause this.
Hi , thanks for the response the extension is enabled on the extension. is there anything else that would cause this issue ?
We have NAT running and like i said it works until it drops .
That’s how SIP register works. It is only about telling the system that receives the register how to reach the system that sends it. It doesn’t stop calls going the other way, as long as they can be authenticated.
I don’t know to what extent FreePBX gives access to type=peer or type=friend, assuming you are using chan_sip, but Asteirisk authenticates type=peer by either the user part of the From header, or the IP address. The IP address is set by REGISTER, but the From address is part of the static configuration. (Best practice on pure Asterisk is not to use type=friend, but I don’t know if you can do this on FreePBX. That will stop calls in both directions, although the real reasons for doing it are different.)
If you want a faster recovery, you will need to reduce the interval between REGISTER requests.