[Asterisk-Users] Cisco and Asterisk 2621
Skuse, Phil
Phil.Skuse at vicorp.com
Wed Dec 3 11:10:53 MST 2003
I have a 2621 working with asterisk. See below:
sip.conf
======
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[cisco] ; Cisco 2621 Router
type=friend
canreinvite=no
insecure=yes
host=192.168.62.1 ; address of the cisco router
dtmfmode=inband
context=default
extensions.conf
===============
; My asterisk numbers are 600-699 (omitted from example)
; Send all calls prefixed with 9 to the cisco
exten => _9.,1,Dial,sip/BYEXTENSION at cisco
relevant part of cisco configuration
====================================
[c2600-is-mz.122-13.T.bin]
!
dial-peer voice 6 pots
description Incoming Call from PSTN to number 6xx
application session
incoming called-number 6..
destination-pattern 6..
no digit-strip
direct-inward-dial
port 1/0:15
!
dial-peer voice 600 voip
description Outgoing call to Asterisk Server for numbers 6xx
application session
destination-pattern 6..
session protocol sipv2
session target ipv4:192.168.62.60
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 9 voip
description Incoming Call from Asterisk Server to number beginning with 9
application session
incoming called-number 9T
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 900 pots
description Outgoing call to PSTN for numbers beginning with 9
application session
destination-pattern 9T
no digit-strip
port 1/0:15
!
-----Original Message-----
From: Ariel Batista [mailto:abatista at avionica.com]
Sent: 03 December 2003 17:06
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Cisco and Asterisk 2621
Ok here is a question that has gotten me stumped. I have an Asterisk system
up and running. I need to connect it via the Internet to a Sip Cisco
system. This is what they have. I have there IP address's and login.
X-lite is able to connect to them and make a call! So I have the name right!
CISCO router model: 2621
VoIP module: NM-HDA-4FXS
I have done Google lookup and at the Wiki about this. What I did get is the
following from them. Following in the SIP.CONF file.
register => name at 217.XXX.XXX.XXX:5060
This does not seem to work.
I have also tried at the extensions.conf a setting of.
exten => 380,1,Dial(SIP/name at 217.XXX.XXX.XXX)
I feel I have missed something some place or I just don't understand what to
do!
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