Asterisk & Cisco Callmanager Express (CME)

I decided to add some voicemail functionality to the lab. The Open Source Project Asterisk came to my mind. The goal was to get the voicemail forwarded, with an audio file, to a public email account at GMX. I also wanted to use the MWI function of the SCCP phones. And surprisingly, after "a little bit" of testing it worked. Remember, this is only a proof of concept, I don't claim that to be the perfect solution. Suggestions welcome.

With help from:http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
Here is my extensions.conf:


root@home-nas:~# cat /etc/asterisk/extensions.conf
[general]

[cme]
exten => 105,1,VoiceMailMain(),p
exten => 106,1,NoOp,${CALLERID(num)}
exten => 106,2,NoOp,${CALLERID(rdnis)}
exten => 106,3,Playback(silence/1)
exten => 106,4,Voicemail(${CALLERID(rdnis)},b)
exten => 106,5,Hangup
exten => 106,106,Hangup

;Transfer on busy.
;see notes above, just sets the b flag for the voicemail application to stat the call was busy (as apposed to unavailable).
exten => 107,1,NoOp,${CALLERID(num)}
exten => 107,2,NoOp,${CALLERID(rdnis)}
exten => 107,3,Playback(silence/1)
exten => 107,4,Voicemail(${CALLERID(rdnis)},u)
exten => 107,5,Hangup
exten => 107,106,Hangup

[vm]

;CCME Specific VM
;Voice mail Key on 79xx - need to use the last 3 digits of the CallerID. See notes on "calling-number local secondary" in the telephony-service section
;of the cisco config
;exten => 105,1,NoOp,${CALLERID(num)}
;exten => 105,2,Background(silence/1)
;exten => 105,3,VoicemailMain(${CALLERID(num)}@default)
;exten => 105,4,Hangup
;exten => 105,104,Hangup
exten => 105,1,VoiceMailMain(${CALLERID(num)})

;Transfer on unavailable.
; I playback 1 second of silence to allow the call to establish correctly else the start of the audio gets cut off, if you have silence suppression or something
; I guess you could play a beep.
; Because the call is being transfered the variable ${CALLERIDNUM} contains the number of the calling device not the divice they were calling
; This would mean you would end up in your own or a non existant mailbox, the variable ${RDNIS} contains the number
; the call was redirected from and therefore can be used to specify the correct mailbox number.
exten => 106,1,NoOp,${CALLERID(num)}
exten => 106,2,NoOp,${CALLERID(rdnis)}
exten => 106,3,Playback(silence/1)
exten => 106,4,Voicemail(${CALLERID(rdnis)},b)
exten => 106,5,Hangup
exten => 106,106,Hangup

;Transfer on busy.
;see notes above, just sets the b flag for the voicemail application to stat the call was busy (as apposed to unavailable).
exten => 107,1,NoOp,${CALLERID(num)}
exten => 107,2,NoOp,${CALLERID(rdnis)}
exten => 107,3,Playback(silence/1)
exten => 107,4,Voicemail(${CALLERID(rdnis)},u)
exten => 107,5,Hangup
exten => 107,106,Hangup

root@home-nas:~#

My sip.conf:

root@home-nas:~# cat /etc/asterisk/sip.conf
[general]
insecure=port,invite
bindport=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
context=cme
host=172.20.29.90
type=peer
;callerid=Unknown

[10]
insecure=port,invite
context=vm
type=friend
qualify=yes
mailbox=10
host=172.20.29.90
;ipaddr=172.20.29.90
dtmfmode=auto
dial=SIP/10
canreinvite=no
callerid=device <10>
subscribemwi=no
nat=no
fromuser=105
vmexten=105
username=10

[53]
insecure=port,invite
context=vm
type=friend
qualify=yes
mailbox=53
host=172.20.29.90
;ipaddr=172.20.29.90
dtmfmode=auto
dial=SIP/53
canreinvite=no
callerid=device <53>
subscribemwi=no
nat=no
fromuser=105
vmexten=105
username=53

root@home-nas:~#

I forwad the emails to a public mail service from GMX.
My voicemail.conf:

root@home-nas:~# cat /etc/asterisk/voicemail.conf
[general]
emailbody=${VM_NAME},\n\New message in mailbox!! ${VM_MAILBOX}\n\n\tVon:\t${VM_CALLERID}\n\tLaenge:\t${VM_DUR} Sekunden\n\tDate:\t${VM_DATE}\n\Dial 105 to access your voicemail.\n\You can also listen at http://172.20.29.88/html/recordings/index.php , to your messages.\n\nAsterisk Rulez!!\n
serveremail=yaya@gmx.net ; Who the e-mail notification should appear to come from
fromstring=Voicemail System
nextaftercmd=yes
externnotify=chmod -R a+rw /var/spool/asterisk/voicemail/default/
[default]
10 => 1234,AJ10 Mailbox,aj10@gmx.de,,attach=yes|saycid=yes|envelope=yes|delete=no
53 => 1234,Andre Mailbox,andre@gmx.de,,attach=yes|saycid=no|envelope=yes|delete=no


root@home-nas:~#

Nokia Phones & Cisco Callmanager Express (CME)

After reading that Nokia phones come with a SIP client, I was curios about testing them with a Cisco Callmanager Express (CME). I have to say it works quite well. I have tested this with Nokia E60, E61, E70 & N95-8GB. MWI doesn't work with the Nokias. This is kind of a braindump of the configuration. I assume that you have a working wireless connection to your LAN. Furthermore I added a Link to a PSTN and to a SIP provider. When I add a 9 as a prefix to the phone number the system will do a dialout to the PSTN, when I dial the number without a special prefix, it will use the SIP Provider. Irrelevant parts of the configuration are partly not shown here. I really recommend the IOS ipvoicek9-mz.124-15.T3.bin for this setup. Comments welcome!!


LAB Configuration:



Phone Configuration:

SIP settings
- New SIP profile -> Use default profile
-- Profile name -> Lab
-- Service profile -> IETF
-- Default access point -> Lab
-- Public user name -> sip:10@172.20.29.90
-- Use compression -> No
-- Registration -> Always on
-- Use security -> No
-- Proxy server
--- Proxy server address -> sip:172.20.29.90
--- Realm -> None
--- Proxy server address -> sip:172.20.29.90
--- User name -> 10@172.20.29.90
--- Password -> xxx
--- Allow loose routing -> yes
--- Transport type -> UDP
--- Port -> 5060
-- Registrar server
--- Registrar server address -> sip:172.20.29.90
--- Realm -> None
--- User name -> 10@172.20.29.90
- --Password -> xxx
--- Allow loose routing -> yes
--- Transport type -> UDP
--- Port -> 5060

Internet tel.
- New profile
-- Name -> Lab
-- SIP profiles -> Lab




CME Configuration:

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.04.22 19:39:16 =~=~=~=~=~=~=~=~=~=~=~=

test_gw#sh run
Building configuration...

Current configuration : 19750 bytes
!
! Last configuration change at 20:42:08 MESZ Tue Apr 1 2008 by andre
! NVRAM config last updated at 21:20:42 MESZ Wed Apr 2 2008
!
version 12.4
!
hostname test_gw
!
aaa new-model
!
!
aaa accounting connection H.323 start-stop group radius
!
!
aaa session-id common
memory-size iomem 25
!
voice-card 0
codec complexity medium
!
voice-card 2
codec complexity medium
!
ip dhcp excluded-address 172.20.29.80 172.20.29.254
!
ip dhcp pool home
network 172.20.29.0 255.255.255.0
next-server 172.20.29.15
default-router 172.20.29.90
dns-server 172.20.29.88
domain-name lab.test
option 150 ip 172.20.29.90
option 66 ip 172.20.29.90
option 128 ip 172.20.29.90
!

!
!
no voice call carrier capacity active
!
voice service pots
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server
no call service stop
!
!
!General SIP Configuration
voice register global
mode cme
source-address 172.20.29.90 port 5060
max-dn 12
max-pool 12
timezone 21
time-format 24
date-format D/M/Y
call-forward system redirecting-expanded
create profile sync 0834091551485351
!Configuration for Nokia N95
voice register dn 2
number 10
allow watch
name nn95
no-reg
label NN95
mwi
!
voice register pool 2
id mac 1111.2222.3333
number 1 dn 2
dtmf-relay rtp-nte
username nn95 password cisco
codec g711alaw
!
!
!
voice translation-rule 9
rule 1 /^9/ //
!
voice translation-rule 40
rule 1 /444582$/ /53/
rule 3 /444631$/ /11/
!
voice translation-rule 41
rule 1 /^53$/ /444582/
rule 3 /^11$/ /444631/
!
voice translation-rule 53
rule 3 /^\(..\)$/ /4922012345\1/
rule 5 /^492201234553$/ /53/

!
!
voice translation-profile SIP-in
translate called 53
!
voice translation-profile SIP-out
translate calling 53
!
voice translation-profile PSTN-in
translate called 40
!
voice translation-profile PSTN-out
translate calling 41
translate called 9
!
!
ip tcp path-mtu-discovery
ip ftp username ciscoftp
ip ftp password xxx
!
!
class-map match-any AutoQoS-VoIP-RTP-Trust
match ip dscp ef
class-map match-any AutoQoS-VoIP-Control-Trust
match ip dscp cs3
match ip dscp af31
!
!
policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority percent 70
class AutoQoS-VoIP-Control-Trust
bandwidth percent 5
class class-default
fair-queue
!
!
gw-accounting syslog
!
!
!
!
interface FastEthernet0/0
ip address 10.99.1.1 255.255.255.0 secondary
ip address 172.20.29.90 255.255.255.0
ip accounting output-packets
ip nbar protocol-discovery
speed auto
ntp broadcast
no cdp log mismatch duplex
service-policy output AutoQoS-Policy-Trust
!
ip route 0.0.0.0 0.0.0.0 172.20.29.81
!
voice-port 0/0
!
voice-port 0/1
!
!
dial-peer voice 10 voip
translation-profile incoming SIP-in
translation-profile outgoing SIP-out
service session
destination-pattern 0.T
session protocol sipv2
session target dns:sip.Provider.SIP
incoming called-number 4922012345..
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
dial-peer voice 50 voip
translation-profile incoming PSTN-in
translation-profile outgoing PSTN-out
destination-pattern 9T
redirect ip2ip
session target ipv4:172.20.29.89
incoming called-number 444[56][839][214]
codec g711alaw
no vad
!If you register SIP phones or SCCP phones, this will give you presence information in the telephony directory
presence
presence call-list
watcher all
allow subscribe
!
!
dial-peer voice 100 voip
destination-pattern ^10[567]$
rtp payload-type nte 98
session protocol sipv2
session target ipv4:172.20.29.88
dtmf-relay rtp-nte
codec g711alaw
no vad
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username 492201234553 password xxx realm Provider.SIP
registrar dns:sip.Provider.SIP:5060 expires 5000
sip-server dns:sip.Provider.SIP
presence enable
!
!
telephony-service
no auto-reg-ephone
load 7960-7940 P00308000400
load ATA ATA030100SCCP040211A.zup
load 7970 SCCP70.8-2-2SR1S
max-ephones 16
max-dn 150
ip source-address 172.20.29.90 port 2000
auto assign 1 to 24
timeouts interdigit 5
time-zone 23
time-format 24
date-format dd-mm-yy
max-conferences 4 gain -6
moh en_bacd_music_on_hold.au
web admin system name webadmin secret xxx
dn-webedit
transfer-system full-blind
secondary-dialtone 0
directory last-name-first
directory entry 1 1234# name Test1
create cnf-files version-stamp 7960 Apr 01 2008 20:13:39
!
!
ephone-dn 1 dual-line
number 53 secondary 492201234553 no-reg primary
pickup-group 5
label 53
description Andre Janssen
name Andre Janssen
allow watch
huntstop channel
no huntstop
mwi sip
!
!
!
!
ephone 1
mac-address 1212.2323.3434
username "andre" password xxx
type 7970
button 1:1
pin 12345
!
!
!
line con 0
password xxx
line aux 0
transport preferred telnet
flowcontrol hardware
line vty 0 4
exec-timeout 180 0
password xxx
!
ntp master 3
ntp server 192.53.103.103
end

test_gw#