Question: SIP VOIP phone getting unregistered from SIP Server after 120 seconds. The 120 seconds is a value of Time to Live (TTL) even when I increase the TTL time to a crazy number this does not help.
Any idea on what I should be looking at?
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Answer: The Unregistering of SIP VOIP phones from a SIP Server could be an indication of something else. Below are what you should be looking at:
1. Check if your Network Firewall is blocking any SIP #Protocol traffic, keep in mind that if you have different other Network Interfaces with #VLAN Configured then you should have a quite complex Network to diagnose.
- After checking the Firewall logs and all is cleared that all SIP traffic is getting In and Out of the Interface then proceed to number 2.
2. Check that the VOIP Phones are located on the same Subnet e.g. If the SIP Server is located on 192.168.1.0, make sure the Phones are also connected to the 192.168.1.0 interface/subnet.
- If this is not the case, then look into the Router responsible for NATing the Traffic to the Phones from the SIP Server.
- If there is a separate Router responsible for Nat'ing the Traffic to the Phones from the SIP Server then go to the WAN Section of the Router and look for the "NAT Passthrough" tab.
- Allow SIP Passthrough to "Enable"
- When this setting is enabled the Phones will be disconnected from the Network and attempt to connect back.
3. If the number 2 did not solve your problems, then look into the SIP.conf file and make sure that all is set correctly.
+1 888-688-2939 
[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=yes
nat=yes;If you have NAT enabled on the Router responsible for Routing traffic to phones
session-timers=refuse
localnet=192.168.0.0/255.255.0.0;This is the Network the Phones are located at
externip=192.168.1.987;This is the SIP Server
pedantic=no;I don't know what this means Added to avoid phone dropouts
[7001]
type=friend
host=dynamic
secret=123
context=internal
qualify=yes;This will allow SIP server to make sure the phone is up
