No NAT involved. But there is a router and the Asterisk box is on a private
IP.
-----Original Message-----
From: George Skorup (Cyber Broadcasting) via Af
Sent: Wednesday, September 17, 2014 4:40 PM
To: af@afmug.com
Subject: Re: [AFMUG] OT Asterisk question
Isn't unregistered SIP called peer-SIP? That's what VoipInnovations'
trunks are. We have put them into Asterisk boxes and it works fine. If
you're dealing with NAT, your NAT/router/firewall box probably has a SIP
helper function that rewrites the SIP messages. And/or the switch might
have NAT traversal that can figure everything out for you.
On 9/17/2014 5:25 PM, Chuck McCown via Af wrote:
Does a register string always have to be populated in trunks?
I am doing a new system connected to a Genband/Nortel switch. And even
though it is our switch, the switch techs here have not done this before.
Genband has not been much help either.
Calls will actually come in, ring an extension but no audio cut through
and the caller will receive fast busy after about 10 seconds. Outbound
calls on that same trunk go to busy.
Debug shows:
"chan_sip.c: Unable to create/find SIP channel for this INVITE" on some of
the calls.
I am thinking this may be an IP problem on the outbound calls. Do not
recall seeing that on the inbound calls, just no audio cut through.
