Cisco Call Manager Express – SIP/SCCP Configuration
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.
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#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#
R1#
R1#
R1#
R1#
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
R1#
R1#
Happy Labing!!
I am working in an IT company and having 10+ years of experience into Cisco IP Telephony and Contact Center. I have worked on products like CUCM, CUC, UCCX, CME/CUE, IM&P, Voice Gateways, VG224, Gatekeepers, Attendant Console, Expressway, Mediasense, Asterisk, Microsoft Teams, Zoom etc. I am not an expert but i keep exploring whenever and wherever i can and share whatever i know. You can visit my LinkedIn profile by clicking on the icon below.
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