Pavel Litvinenko wrote:

Joseph Finley wrote:

I'm not sure if I am wording this correctly, but I'll try.

I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial out. I
guess it's all syntax that I'm doing wrong. Does someone have a couple
small snip-its to accomplish this?


This is what a buddy of mine uses to call my pbx extensions.
!
voice service voip
h323
sip
 bind all source-interface FastEthernet0/0   <<--- the public IP interface
!
! The Cisco 7960s only do these two codecs. (also g711alaw)
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
! An analog port on the 3725 router.
voice-port 1/0/0
description POTS Test Phone
!
! The local number for the analog port.
dial-peer voice 100 pots
application session
destination-pattern 6110
port 1/0/0
!
! forward anything 6XXX to my pbx at 1.2.3.4
dial-peer voice 111 voip
preference 1
destination-pattern 6...
voice-class codec 1
voice-class h323 1
session protocol sipv2
session target ipv4:1.2.3.4
ip qos dscp cs5 media
no vad
!
! I believe this just tells where the server is, it doesn't REGISTER.
sip-ua
sip-server ipv4:1.2.3.4
!


Newer Cisco IOS is supposed to be able to register via SIP, but the version my buddy is running doesn't currently support it.

But he is able to dial my pbx easily, and I can setup the sip.conf with a default
ip for his router, etc.


-Andrew

_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to