Configuration example:
The CME configured as below:
ip dhcp excluded-address 192.168.255.1 192.168.255.10
ip dhcp pool ephone255
network 192.168.255.0 255.255.255.0
default-router 192.168.255.2
option 150 ip 192.168.255.2
!-- option 150 ip must match the ip specified by "ip source-address" under telephony service
lease 7
!
!
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
signaling forward unconditional
sip
bind control source-interface FastEthernet1/0
bind media source-interface FastEthernet1/0
registrar server
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
codec preference 6 g723ar63
!
!
!
!
voice register global
mode cme
source-address 192.168.255.2 port 5060
max-dn 10
max-pool 10
authenticate realm local
timezone 47
date-format D/M/Y
dst start Oct week 1 day Sun time 02:00
dst stop Apr week 1 day Sun time 02:00
tftp-path flash:
file text
create profile sync 0001324321105938
!
voice register dn 1
number 1003
name Peter
label Peter
!
voice register pool 1
id mac 000C.2917.A83B
number 1 dn 1
dtmf-relay rtp-nte
voice-class codec 1
username 1003 password test123
!
!
!
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface FastEthernet1/0
ip address 192.168.255.2 255.255.255.0
duplex auto
speed auto
!
!
!
no ip http server
no ip http secure-server
!
no cdp log mismatch duplex
!
!
!
!
!
!
control-plane
!
!
dial-peer voice 1 voip
destination-pattern 1...
session protocol sipv2
dtmf-relay sip-notify
codec g711alaw
no vad
!
!
!
!
!
telephony-service
max-ephones 190
max-dn 300
ip source-address 192.168.255.2 port 2000
cnf-file location flash:
mwi relay
max-conferences 8 gain -6
web admin system name cisco password cisco
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 1002
label frank
name frank
!
!
ephone 1
device-security-mode none
mac-address 000C.292B.C840
type CIPC
button 1:1
!
!
Troubleshooting
1 . debug voice ccapi inout
The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the router.
This command shows how a call flows through the system. Using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs.
you can dial, and hear the ringing, but it disconnect once you click answer.
2. debug ccsip
- debug ccsip calls: This command displays all SIP call details as they are updated in the SIP call control block. You can use this debug command to monitor call records for suspicious clearing causes.
- debug ccsip errors: This command traces all errors that are encountered by the SIP subsystem.
- debug ccsip events: this command traces event, such as call setups, connections and disconnections. An events version of a debug command is often the best place to start because detailed debugs provide much useful information.
Enable debug:
debug ccsip calls
Redial and generate debug messages:
*Mar 1 00:48:58.203: //27/EFD853388035/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x66BA6E24 State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 1002 Called Number : 1003 Source IP Address (Sig ): 192.168.255.2 Destn SIP Req Addr:Port : 192.168.255.129:5060 Destn SIP Resp Addr:Port : 192.168.255.129:5060 Destination Name : 192.168.255.129 CME2# *Mar 1 00:48:58.203: //27/EFD853388035/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : No Codec //issue here Negotiated Codec Bytes : 0 Negotiated Dtmf-relay : 0 Dtmf-relay Payload : 0 Source IP Address (Media): 192.168.255.2 Source IP Port (Media): 16946 Destn IP Address (Media): 0.0.0.0 Destn IP Port (Media): 0 Orig Destn IP Address:Port (Media): 0.0.0.0:0 *Mar 1 00:48:58.207: //27/EFD853388035/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 65 Disconnect Cause (SIP) : 200
We found there is no codec configured, to resolve this issue, create codec class:
voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 codec preference 4 g729br8 codec preference 5 g722-64 codec preference 6 g723ar63
Apply to Voice register pool:
voice register pool 1 id mac 000C.2917.A83B number 1 dn 1 dtmf-relay rtp-nte voice-class codec 1
Redial and check the debug.
*Mar 1 01:20:14.723: //33/442A29208043/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x66BAB1DC State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 1002 Called Number : 1003 Source IP Address (Sig ): 192.168.255.2 Destn SIP Req Addr:Port : 192.168.255.129:5060 Destn SIP Resp Addr:Port : 192.168.255.129:5060 Destination Name : 192.168.255.129 CME2(config-class)# *Mar 1 01:20:14.727: //33/442A29208043/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : g711alaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay : 6 Dtmf-relay Payload : 101 Source IP Address (Media): 192.168.255.2 Source IP Port (Media): 19492 Destn IP Address (Media): 192.168.255.129 Destn IP Port (Media): 5062 Orig Destn IP Address:Port (Media): 0.0.0.0:0 *Mar 1 01:20:14.731: //33/442A29208043/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 200
SCCP registration
Go to the status message check which tftp file it tried to download.
Make sure the ip address in the DHCP option 150 matches the ip source-address in telephony service configuration mode.
now the two phones should connect
Reference
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/99946-cme-sip-guide.html
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