Cisco Call Manager Express – SIP/SCCP

Configuration
While testing further i had a thought of preparing a lab scenario where i have SCCP Phones
and SIP Phones registered in the same CME and will initiate a call within the lab scenario.
PS:- I have implemented the scenario using GNS3 and can confirm that calls are working
internally from SIP to SCCP and vice versa.

Below is the configuration on the router:

R1#sh run
Building configuration…
Current configuration : 2329 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname R1
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
memory-size iomem 5
no ip icmp rate-limit unreachable
ip cef
!
!
no ip domain lookup
!
multilink bundle-name authenticated
!
!
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 expires max 500 min 60
!
!
voice class codec 10
codec preference 1 g711ulaw
!
!
voice register global
mode cme
source-address 10.1.1.50 port 5060
max-dn 20
max-pool 20
authenticate register
authenticate realm local
create profile sync 0001365470207527
!
voice register dn 1
number 81011000
allow watch
!
!
voice register pool 1
id mac 0000.0000.0000
number 1 dn 1
presence call-list
username 81011000 password 81011000
codec g711ulaw
!
!
archive
log config
hidekeys
!
!
ip tcp synwait-time 5
!
!
interface FastEthernet0/0
ip address 10.1.1.50 255.255.255.0
duplex auto
speed auto
!
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip default-gateway 10.1.1.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.1.1.1
!
!
no ip http server
no ip http secure-server
!
!
control-plane
!
!
dial-peer voice 1 voip
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
!
!
gateway
timer receive-rtp 1200
!
sip-ua
!
!
telephony-service
max-ephones 10
max-dn 10
ip source-address 10.1.1.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 81011002
!
!
ephone 1
device-security-mode none
mac-address 0000.0000.0010
type CIPC
button 1:1
!
!
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
!
!
end
R1#
R1#sh ephone registered
ephone-1 Mac:0000.0000.0010 TCP socket:[1]
activeLine:0 REGISTERED in SCCP ver 9 and Server in ver 8
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0
debug:0 caps:7
IP:10.1.1.2 3872 CIPC keepalive 21 max_line 8
button 1: dn 1 number 81011002 CH1 IDLE
R1#sh sip-ua status registrar
Line destination expires(sec) contact
call-id
peer
============================================================
81011000 10.1.1.2 53 10.1.1.2
Mjg5YjAzNzk2YzUwZGE3NzczNDJkZTI1NDFmZjc4ZWQ.
40001
R1#sh sip-ua calls
SIP UAC CALL INFO
Call 1
SIP Call ID : BA3100C0-2BE911D6-80D6D6C9-
A49469E@10.1.1.50
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 81011002
Called Number : 81011001
Bit Flags : 0xC04018 0x100 0x80
CC Call ID : 121
Source IP Address (Sig ): 10.1.1.50
Destn SIP Req Addr:Port : 10.1.1.4:57383
Destn SIP Resp Addr:Port: 10.1.1.4:57383
Destination Name : 10.1.1.4
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 121
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.1.1.50:17212
Media Dest IP Addr:Port : 10.1.1.4:4004
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Number of SIP User Agent Server(UAS) calls: 0

Happy Labing!!