Using my 2801 + FXO

Joined: Fri 17 of Apr, 2009

Using my 2801 + FXO

Posted:Fri 17 of Apr, 2009 (15:14 UTC)
Hi all, first time post here so please be gentle.

I have a cisco 2801 running CCME v4.1, I would like Asterisk to use my 2801 as the gateway, the reason I want to use Asterisk as well as CCME is so that I can use Nagios to create voice alerts to mobiles, I will post my CCME config below but first will show my IP's so you can see how it hangs together.

Cisco2801 IP address
Asterisk IP address

I'm using the Gui for Asterisk as I'm completely new to this and hoping it will help me out, OK here is what I have on my Router so far

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol pass-through g711alaw
h225 timeout tcp call-idle never
h245 tunnel disable
h245 caps mode restricted
modem passthrough nse codec g711alaw

voice-port 0/1/0
supervisory disconnect anytone
cptone GB
connection plar opx 1005
impedance complex5
description ***POTS LINE 1***
caller-id alerting dsp-pre-allocate

dial-peer voice 11 pots
description *** Outgoing calls ***
translation-profile outgoing 9
preference 1
destination-pattern 9T
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind connect enable 8
progress_ind disconnect enable 8
port 0/1/0

dial-peer voice 2 voip
description Route calls starting with 2 to the Asterisk PBX
destination-pattern 2...
session protocol sipv2
session target ipv4:
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
clid strip
no vad

timer receive-rtp 1200

authentication username asterisk password 13161B1D1F181D7B79
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry options 0
timers trying 1000
sip-server ipv4:

Then from my Asterisk Gui I selected Trunk from the menu on the left hand side, then VOIP Trunks Add new SIP/IAX Trunk.
In the provider name I have Cisco 2801
Username asterisk
Password asterisk
Insecure very

Under System status I see my newly added trunk but it just says Request in Red, below is an output from my debug ccsip, I have somebody can point me in the correct direction

Sorry for the long post on my first post


SIP/2.0 100 Trying
Via: SIP/2.0/UDP;branch=z9hG4bK0ebc892f;rport
From: <sip:asterisk@>;tag=as240cf1c9
To: <sip:asterisk@>
Date: Fri, 17 Apr 2009 14:41:45 GMT
Call-ID: 38b1644e1699186c40f2707d70ad248b@
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0