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This is the support site for Andrews & Arnold Ltd, a UK Internet provider. Information on these pages is generally for our customers but may be useful to others, enjoy!

VoIP Phones - Asterisk: Difference between revisions

VoIP Phones - Asterisk (view source)

Revision as of 21:11, 12 March 2025

115 bytes added ,  12 March
→‎PJSIP: NAT Issues: Keep-Alive / Anti-Idle
 
 
==PJSIP: NAT Issues: Keep-Alive / Anti-Idle==
If you are using a firewall or NAT router with short timeouts on UDP sessions youthe canexample forceregistration configuration above enables OPTIONS packets to be sent over the connection to keep it alive.
A symptom of NAT issues can be inbound call delivery working for a bit after startup, then not working for a bit (with the code "183" in the CDR in the control panel) then calls working again a number of minutes later without you doing anything (or possibly reloading or restarting asterisk will solve it for a bit).
 
As Andrews & Arnold have multiple call servers it is possible callsfor cancall signalling to come from multiple IP addresses. Your NAT session or SIP helper/ALG may only be allowing SIP traffic from the serverIP address your Asterisk registered to.
 
In the PJSIP registration example aboveon this page the problem is mitigatedsolved by having three AOR (Address of Record) sections. All three have qualify_frequency set (this makes asterisk send keepalive packets). The endpoint section then references the 3 aor entries.<br />
Your Asterisk server will send outbound packets to the shared hostname, and the two call servers specifically. This has the effect of keeping the NAT session on your router alive for all the call servers meaning it doesn't matter which call server delivers the INVITE (call), it should always reach your Asterisk server.
 
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