Still no joy but it changed things.
Now, on outbound calls, no busy. Just nothing.
Inbound still rings the extension, no audio, caller hears fast busy after
about 10 seconds.
Will work on IP routing issues tomorrow. Thanks for the suggestions.
-----Original Message-----
From: George Skorup (Cyber Broadcasting) via Af
Sent: Wednesday, September 17, 2014 4:51 PM
To: [email protected]
Subject: Re: [AFMUG] OT Asterisk question
Maybe this helps?
Type=peer is used for outbound trunks. Type=user is used for inbound
trunks. Type=friend allows both inbound and outbound to work on one trunk.
On 9/17/2014 5:43 PM, Chuck McCown via Af wrote:
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: [email protected]
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.
