VoIP Phones - Asterisk: Difference between revisions

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==PJSIP: NAT Issues: Keep-Alive / Anti-Idle==

If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.

A symptom of NAT issues can be inbound call delivery working, then not working for a bit (with the code "183" in the CDR in the control panel) then calls working again 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 calls can come from multiple IP addresses. Your NAT session or SIP helper/ALG may only be allowing SIP traffic from the server your Asterisk registered to.

Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds.

An example of the aor section follows:

In the PJSIP registration example above this is mitigated 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 />

[aaisptrunk]

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.

type = aor

contact = sip:+442082881111@voiceless.aa.net.uk

qualify_frequency=20

==PJSIP: IPv6==