[Asterisk-Users] * to Cisco router with FXO's via SIP
Henry Devito
hdevito at qwest.net
Wed Oct 6 17:16:40 MST 2004
FXS (Foreign Exchange Station) provide dial tone. They are the station
side. FXO (Foreign Exchange Office) Accept Dial tone from a analog phone
line. FYI Here is a working config for using FXO ports in a Cisco router to
connect to asterisk.
////sip.conf/////
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
context=calls2
[192.168.254.100]
context=pstn-incoming
type=friend
host=192.168.254.100 ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
[5000]
context=localphones
type=friend
username=5000
secret=secret
host=dynamic
mailbox=5000
;Henry Soft Phone
[5005]
type=friend
secret=henry
auth=md5
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
callerid="Henry Devito" <5005>
allow=gsm
allow=ulaw
allow=alaw
context=default
////END sip.conf/////
////extensions.conf/////
[calls2]
exten => _.,1,Congestion
[pstn-incoming]
include => lan-phones
[local-phones]
include => lan-phones
include => pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to the
PSTN
;exten => _9.,1,Dial(SIP/${EXTEN:1}@192.168.254.100) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
exten => _9XXXX,1,Dial,SIP/${EXTEN}@192.168.254.100 ; IP address of Cisco
gateway
exten => _9XXXX,2,Congestion
[lan-phones]
exten => 5005,1,Dial(SIP/5005,20)
exten => 5005,2,Voicemail(u5005)
exten => 5005,102,Voicemail(b5005)
exten => 5005,103,Hangup
////End Extensions.conf////
////Router config/////
clock timezone GMT 0 ---This is important the router uses GMT for SIP
voice-port 1
cptone DK
connection plar 5005
voice-port 2
cptone DK
connection plar 5006
!
voice-port 3
cptone DK
connection plar 5007
!
voice-port 4
cptone DK
connection plar 5008
!
!
dial-peer voice 100 pots
destination-pattern 9.......
port 1
forward-digits 7
dial-peer voice 2 voip
description Route calls starting with 5 to the Asterisk PBX
destination-pattern 5...
session protocol sipv2
session target ipv4:192.168.254.20:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.254.20
-----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 7:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] * to Cisco router with FXO's via SIP
On Wed, 2004-10-06 at 19:37, Henry Devito wrote:
> Meant to ask you. What type of router is this?
Cisco 827-4v
> Which FXO module do you have
> in the Router?
? It's got FXS's on it. I get confused by that. So that means it does
VOIP to analog. Not analog to VOIP? Arrgghh
So I guess I will need a Digium card FXO or something.
> What is the current IOS level?
Cisco Internetwork Operating System Software
IOS (tm) C820 Software (C820-OV6Y6-M), Version 12.2(15)T, RELEASE SOFTWARE
(fc1 )
TAC Support: http://www.cisco.com/tac
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Tue 11-Mar-03 18:15 by ccai
Image text-base: 0x80013148, data-base: 0x80B9B00C
ROM: System Bootstrap, Version 12.2(1r)XE2, RELEASE SOFTWARE (fc1)
adsl uptime is 39 minutes
System returned to ROM by power-on
System image file is "flash:c820-ov6y6-mz.122-15.T.bin"
CISCO C827-4V (MPC855T) processor (revision 0x502) with 48128K/1024K bytes
of memory.
Processor board ID JAD04380OUW (346214163), with hardware revision 1987
CPU rev number 5
Bridging software.
4 POTS Ports
1 Ethernet/IEEE 802.3 interface(s)
1 ATM network interface(s)
128K bytes of non-volatile configuration memory.
8192K bytes of processor board System flash (Read/Write)
2048K bytes of processor board Web flash (Read/Write)
Configuration register is 0x2102
--
Sincerely,
William L. Thomson Jr.
Support Group
Obsidian-Studios, Inc.
http://www.obsidian-studios.com
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list