NAT settings to resolve disconnections

NAT settings to resolve disconnections

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 - NAT settings to resolve disconnections

Elastix – NAT settings to resolve disconnections

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
externip =
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 [19172]: chan_sip.c: 3551 retrans_pkt: Retransmission timed out in 1a9131e90ba3eab20c60485e6e23caa0 transmission @ XXXX: 5060 for seqno 102 (Critical Request) –



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