"Connected line update prevented" and "Retransmission timeout" issues

General Help
1

Hello,

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”
“Retransmission timeout”

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.

Thank you

2

bump

3

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

4

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 :slight_smile:

5

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.

6

Actually filtering outbound is always a pain. Why do you need to do that?