OK so I have my asterisk server on the inside of the company. i have a couple of users who work remotely , and the boss wants them to have a company phone.
SO I have attempted to use one up. now I have is connecting and can make calls but;
- The person can hear me but I can’t hear them
- A call can not be placed to that extension
when I do a SIP show peers the extension looks like this
699/699 192.168.1.125 D A 5060 UNREACHABLE
In rtp.conf I have tightened up the UDP ports to 11000-12000 and opened that up on my firewall along with Port 5060.
what I have I missed ?
If the remote phone is connected via a VPN, you need to add the VPN subnet to Local Networks (in Asterisk SIP Settings).
Otherwise, you need to set nat=yes for the remote extension.
If you had already done the above, or it does not help, please provide details about your setup:
At remote end: Phone make/model? NAT settings? Router make/model? VPN on? Modem type? Is it configured as a router?
At your end, does Asterisk have a public IP address? Firewall have a SIP ALG?
Btw, do you use FreePBX?
Writing that rtp.conf file directly would likely be overwritten by the engine. You should be doing on GUI.
NAT is mostly the culprit for audio issues and it also shows that Asterisk cannot reach to the end point.
No2 is due to that UNREACHABLE thing.
Bingo it was the Nat=yes I was missing , and no I did not edit the rtp.conf directly ( I wanted to , But I know better )
Thank you .