I’m having line issues with one of my remote users.
When the remote user places a call to me, upon pickup the call will drop in 2 seconds.
When I dial out to that same remote user, the line doesn’t ring, it just goes to voicemail.
Looking at the logfiles, i see these two errors:
“Connected line update prevented”
I opened up the RTP audio stream ports 10000 - 20000 on the remote user’s router to see if that helps resolve the issue, but it does not.
I don’t know what’s causing these issues, can I get some help?
I’ll provide more information if you need, just ask what details you need to see.
Are you sure you have those ports open on your server? or your server’s router? they would never need to be opened on the client just the server side
yes, i triple checked to make sure that both the remote router and our firewall have those ports open.
i did end up finding the solution today though. it was in the same vein as RTP transmissions, but it was Peer RTP. i had to use sip debug to find what port the Peer RTP was transmitting for the calls, and it changes every time a call is placed, so i just opened a wide range on the remote machine and the problem was fixed.
the range i opened was 2200 - 2299 TCP/UDP. It didn’t specify, but I’m assuming that it’s UDP just like the other RTP transmission.
Anyways, if anyone else has the same issue, that’s how to fix it
How to really fix it is to have /etc/asterisk/rtp.conf match the ports that you open on the router, you need two ports per concurrent call so 10000-20000 is usually much too big.
Actually filtering outbound is always a pain. Why do you need to do that?