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

[William L. Thomson Jr.]

On Wed, 2004-10-06 at 18:33, Henry Devito wrote:
> Hi,  Hopefully you can get a little more help here.

Thank you thank you thank you.

>   Can you post your cisco
> config and your * extensions.conf.

Sure I will keep the cisco to what's relevant. Doubt you care about the
rest. The rest I will include as an attachment/link.
http://www.obsidian-studios.com/extensions.conf
http://www.obsidian-studios.com/sip.conf


voice-port 1
 cptone DK
!
voice-port 2
 cptone DK
!
voice-port 3
 cptone DK
!
voice-port 4
 cptone DK
!
dial-peer voice 1 pots
 destination-pattern 5111
 port 1
!
dial-peer voice 2 pots
 destination-pattern 5112
 port 2
!
dial-peer voice 3 pots
 destination-pattern 5113
 port 3
!
dial-peer voice 4 pots
 destination-pattern 5114
 port 4
!
dial-peer voice 10 voip
 destination-pattern .+T
 session protocol sipv2
 session target sip-server
 codec g711alaw bytes 80
!
sip-ua
 retry invite 3
 retry cancel 2
 sip-server ipv4:192.168.1.10:5060
!

Although I do need to update that last one to reflect * new IP.
192.168.1.1

>   The example you were looking at is for
> FXS station ports on the cisco not FXO CO ports.

They are what I plug the outside phone lines into no?

>   I am trying to get this
> working in my lab to help you out. 

Great. Really appreciate it. I will post a wiki or something once I get
a working config for others. So they can keep their teeth.


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