Cisco Call Manager Express (CME) – 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#sh sip-ua calls SIP UAC CALL INFO Call 1 SIP Call ID : [email protected] 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
Cisco Call Manager Express (CME) is a robust telephony solution that supports both SIP and SCCP protocols. With its comprehensive configuration options, CME allows for seamless integration of IP phones and provides advanced call management features. Whether you prefer the simplicity of SIP or the versatility of SCCP, Cisco Call Manager Express offers a reliable solution for your communication needs. Happy Labbing!
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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