It has to be related to NAT and/or reinvite.
Look for what IP is being presented to the outside world. There are
definitely relevant settings in Asterisk, could be in the phone too.
Assuming the phones have private IPs, I'd set canreinvite=no for the SIP
peer used by the phones, and make sure the phones are not set to
override the WAN IP.
Asterisk has a setting showing what IP it uses to talk to the
world....I'm sorry I can't recall the name of the option. If it has its
own public IP, then use that. If it's private and behind NAT then it
may need to present to the world that it has the router's WAN IP. The
NAT router could have SIP helper features. You may need to turn that
either on or off depending on what else you have going on.
You'll get better clues from a packet capture. You can use tcpdump on
the Asterisk Box, and/or do it on the router. You can verify whether
the incoming RTP stream is or isn't getting to you. You can check the
outbound SIP packets to see what source IP is being used. The other end
doesn't reply to the source from the IP header, he replies to the source
IP used in the SIP messages. If Asterisk is identifying itself as
[email protected] then your carrier may be sending RTP to the private
address which obviously can't cross the internet.
------ Original Message ------
From: [email protected]
To: [email protected]
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk
I am grateful for any bones that can be tossed this way. Just stating
“hey, I am having this problem” which implies that any suggestions to
fix the problem are most welcome.
