CME - CLI Config
This is a sample CLI configuration for Call Manager Express (CME).

Please make sure to setup all of the local functionality of the CME before trying to setup the SIP Trunks.

When you are ready to setup the SIP trunks, the first thing you wll need to do is setup a translation rule. The translation rule will help you structure how outbound calls are dialed and sent to any 3rd Party SIP provider for trunking services. Read on…

Translation Rule:

voice translation-rule 3
rule 1 /^9\(…….\)$/ /+1626\1/
!– Local Calling “626″ is the local area code
rule 2 /^9\(……….\)$/ /+1\1/
!– 10 Digit Calling adds “+1″
rule 3 /^9\(.*\)$/ /+\1/
!– 11 Digit Dialing adds “+”
rule 4 /^9\(………..\)$/ /+\1/
!– Catch-all
rule 5 /^9011\(.*\)$/ /+\1/
!– International Dialing strips the “011″ and adds “+”

Configuration of a translation-profile that ties a multiple translation-rule behavior

Translation-Profile:
(I’m still testing this. there is a difference between “translation-rule” and “voice translation-rule” commands for voip-to-voip vs. voip-to-pots)

voice translation-profile SIP-OUTBOUND
translate calling 2
translate called 1
translate redirect-target 3
translate redirect-called 3

The Dial-Peer is next. It is where the acctual trunk information is setup.

Dial-Peer:

dial-peer voice 1 voip
description ** Outgoinging call to SIP trunk **
translation-profile outgoing SIP-OUTBOUND
destination-pattern 9[2-9]……T
voice-class codec 1
!– force dial-tones to pass inband in the SIP control channel, otherwise the tones don’t get sent across properly for IVRs –!
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:216.82.224.202
!– This is the IP address of your SIP provider’s server (bandwidth.com, etc) –!
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
clid network-number 6265551212
!– This is how you setup for a global outbound callerID, only if your SIP provider allows –!
no vad

Now here is a sample of how to configure a user phone:

ephone-dn 1 dual-line
number 1212 secondary +16265551212
!– Make sure you insert the “+1″ into the number in order to recognize inbound calls. –!
label 6265551212
description Temp User
name Temp User
call-forward noan 6000
!– timeout 10 This is to call FWD no Answer to VM @ extension 6000 –!
corlist incoming user900-international

SIP User Agent:
Registration to the SIP proxy server –

sip-ua
! — authentication to for SIP registration –!
credentials username password realm
! — authentication for SIP proxy when connecting calls –!
authentication username password realm
no remote-party-id
! — optional: this line below tells your SIP server the caller-id to send to the called number –!
calling-info sip-to-pstn number set
! — optional: what is received on YOUR IP phone’s caller-id display for incoming calls from SIP server –!
! — note: if you set these calling-info commands, it rewrites the caller-id info so you won’t see who’s the original caller –!
calling-info pstn-to-sip from number set
retry invite 2
retry register 10
timers connect 100
registrar dns:sip3.voipvoip.com expires 3600
sip-server dns:sip3.voipvoip.com
notify telephone-event max-duration 500
host-registrar
presence enable