[Asterisk-Users] * to Cisco router with FXO's via SIP

Henry Devito hdevito at qwest.net
Wed Oct 6 15:33:06 MST 2004


Hi,  Hopefully you can get a little more help here.  Can you post your cisco
config and your * extensions.conf.  The example you were looking at is for
FXS station ports on the cisco not FXO CO ports.  I am trying to get this
working in my lab to help you out. 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of William L.
Thomson Jr.
Sent: Wednesday, October 06, 2004 5:11 PM
To: asterisk-users
Subject: [Asterisk-Users] * to Cisco router with FXO's via SIP

Ok, very frustrated after spending most of the day onthe * irc channel
with little to no help. Mostly just a bunch of crap about being a
newbie, going and reading voip-info.org. etc.

Despite me doing all that already.

My situation is not good but here it is. Hurricane came through, power
spikes killed PBX. Just trying to replace it affordable and possibly
with a few more features.

I am using * v 0.9.0 on Gentoo. Tried going to 1.0.0. Several times. It
just crashes. Bought the wrong phones Cisco 7910 so I am stuck using
SCCP/Skinny.

Got SCCP working with * after loosing a couple teeth. All internal
phones work and except for festival causing some issues. I am ready to
go for the most part.

Final step which I had hoped to complete today was connecting * to the
outside world. I have a Cisco 827-4v with 4 FXO's on it. I have it
configured to use SIP. I used this example

http://www.loligo.com/asterisk/cisco/827-4v/cisco-827-4v-with-asterisk-versi
on1

Despite repeated attempts at asking for help on the * irc channel I made
no progress. I have a sip softphone installed on my laptop and that's
about it. People on the irc channel are just plain rude to newbies. Hope
this is not a reflection of the entire * community?

Basic ?'s

Is * the SIP server or client in my scenario? It seems it's the client
and my router is the server. I was trying to test out the router
directly via a soft phone but that's not really working. Not sure on
syntax. etc. It was recommended to test out the router via a soft phone
on the irc. However once I got the soft phone and it would not connect I
was left out on my own again.

Do I need to have register lines in my sip.conf?
register => user:secret:authuser at host:port/extension

I tried to create an extension for the inside to dial out. Basically I
would like to press 9 get an outside line and dial. Then again at this
point anyway to dial out would be great.

I just want to dial out and get calls in at this point.

exten => 9,1,Dial(SIP/5114)

I am really confused

Please help. Any help is greatly appreciated.

-- 
Sincerely,
William L. Thomson Jr.
Support Group
Obsidian-Studios, Inc.
http://www.obsidian-studios.com

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