After deploying Elastix, it may be necessary to configure NAT settings within asterisk. If you find that your calls are muted, or hang up/disconnect after just a few seconds, you should follow these instructions to configure NAT.
These instructions require you to login to the VM console and edit a text file on the Linux operating system.
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Elastix – Instructions
1. Log in to your Elastix console as “root”
2. Edit the file “/etc/asterisk/sip_nat.conf” (by default this file is empty).
Add the following lines:
nat = yes
localnet = 192.168.0.0 / 255.255.255.0
3. Save the file and reload asterisk with the command “reload asterisk service”
Calls should now proceed normally.NOTE: These instructions may also resolve the following error message seen in the asterisck logs, especially if you are using IP phones behind NAT:
[2013/08/12 15:04:31] WARNING : chan_sip.c: 3551 retrans_pkt: Retransmission timed out in 1a9131e90ba3eab20c60485e6e23caa0 transmission @ XXXX: 5060 for seqno 102 (Critical Request) –