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This document provides step-by-step instructions for setting up a standalone Cisco Unified Communications Manager Express (CME) that uses SIP phones. The document outlines a Cisco Unified Communications Manager Express system with four SIP phones, with configurations for setting up the Cisco Unified Communications Manager Express system and SIP phones. Note: Though the document covers configuration steps to allow Cisco Unified Communications Manager Express to interoperate with Cisco Unity Express, the Cisco Unity Express configuration is outside of the scope of this paper. Refer to Cisco CallManager Express/Cisco Unity Express Configuration Example for more information on Cisco Unified Communications Manager Express and Cisco Unity Express configurations.
The information in this document is based on these firmware versions:
Refer to Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix in order to determine the appropriate SIP firmware to use for each Cisco Unified Communications Manager Express version. Since Cisco Unified Communications Manager Express 4.2 is used, refer to the Cisco Unified Communications Manager Express 4.2 Specifications link.
The SIP phoneloads can be downloaded from these locations:
After you have unzipped both the ZIP files in your TFTP folder, copy all the firmware files onto the Cisco Unified Communications Manager Express flash with your TFTP server. Make sure you copy all these files onto flash.
SIP3951.8-0-2-9.loads SIP3951.8-0-2-9.zz DSP3951.0-0-0-1.zz BOOT3951.0-0-0-9.zz SIP70.8-2-1S.loads term70.default.loads term71.default.loads apps70.8-0-2-55.sbn cnu70.8-2-0-55.sbn cvm70.sip.8-2-0-55.sbn dsp70.8-2-0-55.sbn jar70.sip.8-0-2-25.sbn
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
Refer to the Cisco Technical Tips Conventions for more information on document conventions.
In this section, you are presented with the information to configure the features described in this document.
These tables outline the addressing schemes that are used in this setup.
| Purpose | VLAN | Network | Interface | Interface Address |
|---|---|---|---|---|
| Voice | 192 | 192.168.10.0/24 | VLAN 192 | 192.168.10.1/24 |
| Data | 100 | 10.10.10.0/24 | VLAN 100 | 10.10.10.1/24 |
| Protocol | Phone Type | Extension Number | Phone Number External Mask |
|---|---|---|---|
| SIP | 7970 | 101 | 4085251001 |
| SIP | 7970 | 102 | 4085251002 |
| SIP | 3911 | 103 | 4085251003 |
| SIP | 3911 | 104 | 4085251004 |
| Voicemail Pilot Number | 100 | AA Pilot | 110 |
| MWI On | 800 | MWI Off | 801 |
Note: Use the in order to obtain more information on the commands used in this section.
This document uses this network setup:
This document uses these configurations:
It is necessary to configure two separate DHCP pools; IP Phones use the Voice DHCP pool and PCs use the Data DHCP pool. IP Phones need to use DHCP option 150 in order to provide the IP address of the TFTP Server.
If there are any devices in either pool with static IP addresses, make sure these addresses are excluded from the DHCP pool in order to avoid addressing conflicts. You can use the show ip dhcp binding command in order to verify which addresses the IP Phones and PCs receive from the router.
ip dhcp excluded-address 10.10.10.1 10.10.10.10 ip dhcp excluded-address 192.168.10.1 192.168.10.10 ! ip dhcp pool data network 10.10.10.0 255.255.255.0 default-router 10.10.10.1 ! ip dhcp pool voice network 192.168.10.0 255.255.255.0 option 150 ip 192.168.10.1 default-router 192.168.10.1
In this section, you configure the VLAN interfaces for both the Data and Voice VLAN and assign switchports into their respective VLANs.
Note: Prior to the configuration of VLANs, be sure to add the previous VLANs to the VLAN database with these commands:
CME-SIP#vlan database % Warning: It is recommended to configure VLAN from config mode, as VLAN database mode is being deprecated. Please consult user documentation for configuring VTP/VLAN in config mode. CME-SIP(vlan)#vlan 100 VLAN 100 modified: CME-SIP(vlan)#vlan 192 VLAN 192 modified: CME-SIP(vlan)#exit APPLY completed. Exiting. CME-SIP#
Configure the switchports to be connected to both the Voice and Data VLANs. IP Phones are automatically assigned into the Voice VLAN and PCs connected to either the switchport directly or connected to the switchport on the IP Phone that is assigned to the Data VLAN.
interface FastEthernet0/3/0 description 7970 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/1 description 7970 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/2 description 3911 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/3 description 3911 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! ! interface Vlan100 description Data VLAN ip address 10.10.10.1 255.255.255.0 ! interface Vlan192 description Voice VLAN ip address 192.168.10.1 255.255.255.0
This configuration allows Cisco Unified Communications Manager Express to serve the IP Phones their firmware.
Note: This configuration is mandatory.
tftp-server flash:SIP3951.8-0-2-9.loads tftp-server flash:SIP3951.8-0-2-9.zz tftp-server flash:DSP3951.0-0-0-1.zz tftp-server flash:BOOT3951.0-0-0-9.zz tftp-server flash:SIP70.8-2-1S.loads tftp-server flash:term70.default.loads tftp-server flash:term71.default.loads tftp-server flash:apps70.8-0-2-55.sbn tftp-server flash:cnu70.8-2-0-55.sbn tftp-server flash:cvm70.sip.8-2-0-55.sbn tftp-server flash:dsp70.8-2-0-55.sbn tftp-server flash:jar70.sip.8-0-2-25.sbn
Configure system to allow calls from SIP to SIP endpoints and enable SIP registrar.
Note: This configuration is mandatory.
voice service voip allow-connections sip to sip !--- Enable SIP to SIP calls. sip registrar server expires max 1200 min 300 !--- Enable Cisco IOS SIP registrar.
In this section, you configure voice register global parameters.
Note: Voice Register global configurations for SIP are similar to telephony-service configuration parameters for SCCP phones.
Note: This configuration is mandatory.
voice register global mode cme !--- Set Cisco IOS SIP registrar to CME mode. source-address 192.168.10.1 port 5060 !--- Set the source address for phone registration. max-dn 20 !--- Set max extensions. max-pool 10 !--- Set max phones. load 7970 SIP7 SIP70.8-2-1S !--- Specify phone loads for each phone type. load 3911 SIP3951.8-0-2-9 !--- Specify phone loads for each phone type. authenticate register !--- Set authentication for phone registration. authenticate realm cisco.com tftp-path flash: !--- Specify path for tftp files. create profile !--- Create configuration files for all phones. dialplan-pattern 1 4085251. extension-length 3 !--- Configure dial-plan pattern for the system.
Here is a link to a video on the Cisco Support Community which explains the procedure to register an IP Phone with Cisco Unified Communications Manager Express (CME) using SIP Protocol:
Configure necessary dial-peers and MWI ephone-dns to interoperate with Cisco Unity Express. In order for Cisco Unified Communications Manager Express to interoperate with Cisco Unity Express, it is necessary to configure SIP Cisco Unified Communications Manager Express as a back to back user agent (B2BUA), which means that all the signaling and RTP stream goes through the Cisco Unified Communications Manager Express. This configuration is required in order to enable connectivity to Cisco Unity Express.
dial-peer voice 2 voip destination-pattern 1.0 !--- Specify destination-pattern to reach CUE VM and AA. session target ipv4:10.1.10.1 !--- Configure IP address to reach Cisco Unity Express. session protocol sipv2 dtmf-relay sip-notify !--- Configure DTMF method to communicate with Cisco Unity Express. b2bua !--- Enable B2BUA for Cisco Unified Communications Manager Express !--- for calls to Cisco Unity Express. codec g711ulaw no vad
Configure Cisco Unity Express MWI support for outcall in order to enable MWI for SIP phones.
ephone-dn 11 number 800 mwi on ! ephone-dn 12 number 801 mwi off
Configure voice register dn in order to create extension numbers for ephones. In the previous network topology, there are four extensions, which need to be created as given here.
Note: This configuration is mandatory.
voice register dn 1 name Phone1 !--- Set display name. label 4085251001 !--- Set display label. number 101 !--- Set extension number. call-forward b2bua noan 100 timeout 20 !--- Configure call forward noan to voicemail pilot. call-forward b2bua busy 100 timeout 20 !--- Configure call forward busy to voicemail pilot. allow watch !--- Allow this number to be watched (presence). ! voice register dn 2 name Phone2 label 4085251002 number 102 call-forward b2bua noan 100 timeout 20 !--- Configure call forward noan to voicemail pilot. call-forward b2bua busy 100 timeout 20 !--- Configure call forward busy to voicemail pilot. allow watch ! voice register dn 3 name Phone3 label 4085251003 number 103 call-forward b2bua noan 100 timeout 20 !--- Configure call forward noan to voicemail pilot. call-forward b2bua busy 100 timeout 20 !--- Configure call forward busy to voicemail pilot. allow watch ! voice register dn 4 name Phone4 label 4085251004 number 104 call-forward b2bua noan 100 timeout 20 !--- Configure call forward noan to voicemail pilot. call-forward b2bua busy 100 timeout 20 !--- Configure call forward busy to voicemail pilot. allow watch
Configure voice register pool parameters for each SIP phone.
Note: Voice register pool for SIP phones is identical to ephones for SCCP phones.
Note: This configuration is mandatory.
voice register pool 3 id mac 001A.A11B.500E !--- Specify phone mac-address. type 3911 !--- Specify phone type. number 1 dn 3 !--- Assign button 1 dn tag 3. dtmf-relay sip-notify !--- Configure dtmf-relay sip-notify to work !--- with Cisco Unity Express. codec g711ulaw !--- Specify codec. username user1 password cisco !--- Configure username and password for SIP registrar.
Note: Multiple methods for DTMF can be configured under voice register pool, but for each SIP phone that has a voicemail box on Cisco Unity Express, configure dtmf-relay sip-notify.
In this section, you configure advanced parameters for SIP phones such as presence with Busy Lamp Field (BLF) status. Presence with BLF allows either a SCCP phone or SIP phone to monitor the status of another SIP extensions, which enables presence information between phones.
Note: This is an optional configuration.
These phones support SIP presence service on Cisco Unified Communications Manager Express.
Restrictions
BLF Call-List Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. BLF Speed-Dial Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
Enable Presence for internal lines
Complete these steps in order to enable the router to accept incoming presence requests from internal watchers and SIP trunks.
1. enable 2. configure terminal 3. sip-ua 4. presence enable 5. exit 6. presence 7. max-subscription number 8. presence call-list 9. end
Presence !--- Enable presence service. presence call-list !--- Enable BLF monitoring of directory numbers. max-subscription 120 !--- Configure max number watched sessions. ! sip-ua presence enable !--- Enable router to accept incoming presence request.
Enable a Directory Number to be watched
Complete these steps in order to enable a line associated with a directory number to be monitored by a phone registered to a Cisco Unified Communications Express router. The line is enabled as a presentity and phones can subscribe to its line status through BLF call-list and BLG speed-dial features. There is no restriction on the type of phone that can have its lines monitored; any line on any IP phone or on an analog phone on supported voice gateways can be a presentity.
1. enable 2. configure terminal 3. voice register dn dn-tag 4. number number 5. allow watch 6. end
voice register dn 1 number 101 allow watch !--- Allow this number to be watched. name Phone1 label 4085251001
Note: Repeat this configuration for each extension number that needs to be watched. This step was already done when you first configured voice register dns.
Enable the SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory numbers (presentities) through the BLF speed-dial and BLF call-list presence features. Complete these steps in order to enable the BLF notification features on a SIP phone:
1. enable 2. configure terminal 3. voice register pool pool-tag 4. number tag dn dn-tag 5. blf-speed-dial tag number label string 6. presence call-list 7. exit 8. voice register global 9. mode cme 10. create profile 11. restart 12. end
voice register pool 1 id mac 0016.47CD.9BD7 type 7970 number 1 dn 1 presence call-list !--- Enable this phone to have presence call list. dtmf-relay sip-notify username user1 password cisco codec g711ulaw blf-speed-dial 2 102 label "Phone2" !--- Enable this line to monitor extension 1002. blf-speed-dial 3 103 label "3911-1" !--- Enable this line to monitor extension 1003. blf-speed-dial 4 104 label "3911-2" !--- Enable this line to monitor extension 1004.
Note: Be sure to perform restart every time you change a SIP phone configuration.
Note: Refer to How to Configure Presence Service for more information on the configurations of the SIP Presence Service.
In this section, extensions 102, 103, and 104 are assigned into a parallel hunt group. A parallel hunt group is a hunt group that rings all members in the group simultaneously.
voice hunt-group 1 pilot 180 !--- Configure Hunt group pilot number. list 102, 103, 104 !--- Specify members in hunt-group. final 100 !--- Specify final number as Voicemail Pilot.
This section provides the complete sample configuration for setting up a standalone Cisco Unified Communications Manager Express that uses SIP phones.
CME-SIP#show version Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.4(11)XW2, RELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2007 by Cisco Systems, Inc. Compiled Mon 02-Jul-07 19:10 by prod_rel_team ROM: System Bootstrap, Version 12.3(8r)T6, RELEASE SOFTWARE (fc1) CME-SIP uptime is 18 hours, 55 minutes System returned to ROM by reload at 17:01:34 UTC Wed Oct 3 2007 System image file is "flash:c2801-ipvoice-mz.124-11.XW2.bin" Cisco 2801 (revision 4.1) with 235520K/26624K bytes of memory. Processor board ID FHK084510HS 11 FastEthernet interfaces 1 terminal line 2 Voice FXO interfaces 3 DSPs, 48 Voice resources 1 cisco service engine(s) DRAM configuration is 64 bits wide with parity disabled. 191K bytes of NVRAM. 62720K bytes of ATA CompactFlash (Read/Write) Configuration register is 0x2102 CME-SIP#show running-config Building configuration. Current configuration : 6227 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname CME-SIP ! boot-start-marker boot-end-marker ! logging buffered 999999 no logging console enable password cisco ! no aaa new-model ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 10.10.10.1 10.10.10.10 ip dhcp excluded-address 192.168.10.1 192.168.10.10 ! ip dhcp pool data network 10.10.10.0 255.255.255.0 default-router 10.10.10.1 ! ip dhcp pool voice network 192.168.10.0 255.255.255.0 option 150 ip 192.168.10.1 default-router 192.168.10.1 ! ! no ip domain lookup multilink bundle-name authenticated ! ! ! voice service voip allow-connections sip to sip sip registrar server expires max 1200 min 300 ! ! ! ! ! voice register global mode cme source-address 192.168.10.1 port 5060 max-dn 20 max-pool 10 load 7970 SIP70.8-2-1S load 3911 SIP3951.8-0-2-9 authenticate register authenticate realm cisco.com voicemail 100 tftp-path flash: create profile sync 0000589556325309 ! voice register dn 1 number 101 call-forward b2bua noan 100 timeout 20 allow watch name Phone1 label 4085251001 ! voice register dn 2 number 102 call-forward b2bua noan 100 timeout 20 allow watch name Phone2 label 4085251002 ! voice register dn 3 number 103 call-forward b2bua noan 100 timeout 20 allow watch name Phone3 label 4085251003 ! voice register dn 4 number 104 call-forward b2bua noan 100 timeout 20 allow watch name Phone4 label 4085251004 ! voice register pool 1 id mac 0016.47CD.9BD7 type 7970 number 1 dn 1 presence call-list dtmf-relay sip-notify username user1 password cisco codec g711ulaw blf-speed-dial 2 102 label "Phone2" blf-speed-dial 3 103 label "3911-1" blf-speed-dial 4 104 label "3911-2" ! voice register pool 2 id mac 0014.6948.1D52 type 7970 number 1 dn 2 dtmf-relay sip-notify username user2 password cisco codec g711ulaw ! voice register pool 3 id mac 001A.A11B.4FCE type 3911 number 1 dn 3 dtmf-relay sip-notify username user3 password cisco codec g711ulaw ! voice register pool 4 id mac 001A.A11B.500E type 3911 number 1 dn 4 dtmf-relay sip-notify username user4 password cisco codec g711ulaw ! voice hunt-group 1 parallel final 100 list 102,103,104 pilot 180 ! ! ! ! voice-card 0 ! ! ! archive log config hidekeys ! ! ! interface Loopback0 ip address 10.1.10.2 255.255.255.0 ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 10.1.10.1 255.255.255.0 service-module ip default-gateway 10.1.10.2 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 description 7970 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/1 description 7970 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/2 description 3911 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/3 description 3911 Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/4 description Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/5 description Phone switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/6 description Phone switchport access vlan 192 switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/7 description Phone switchport access vlan 192 switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 192 spanning-tree portfast ! interface FastEthernet0/3/8 switchport access vlan 192 ! interface Vlan1 no ip address ! interface Vlan100 ip address 10.10.10.1 255.255.255.0 ! interface Vlan192 ip address 192.168.10.1 255.255.255.0 ! ip route 10.1.10.1 255.255.255.255 Service-Engine0/0 ! ! ip http server ! ! ! tftp-server flash:BOOT3951.0-0-0-9.zz tftp-server flash:SIP3951.8-0-2-9.zz tftp-server flash:DSP3951.0-0-0-1.zz tftp-server flash:SIP3951.8-0-2-9.loads tftp-server flash:SIP70.8-2-1S.loads tftp-server flash:term70.default.loads tftp-server flash:term71.default.loads tftp-server flash:apps70.8-0-2-55.sbn tftp-server flash:cnu70.8-2-0-55.sbn tftp-server flash:cvm70.sip.8-2-0-55.sbn tftp-server flash:dsp70.8-2-0-55.sbn tftp-server flash:jar70.sip.8-0-2-25.sbn ! control-plane ! ! ! voice-port 0/1/0 ! voice-port 0/1/1 ! ! ! ! ! dial-peer voice 2 voip description ** cue voicemail pilot number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 100 b2bua session protocol sipv2 session target ipv4:10.1.10.1 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 3 voip description ** cue auto attendant number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 110 b2bua session protocol sipv2 session target ipv4:10.1.10.1 dtmf-relay sip-notify codec g711ulaw no vad ! ! presence presence call-list max-subscription 120 ! sip-ua presence enable ! ! telephony-service max-ephones 24 max-dn 72 ip source-address 10.100.100.10 port 2000 system message CME1 time-zone 5 voicemail 100 max-conferences 8 gain -6 call-forward pattern .T web admin system name cisco secret 5 $1$4FC/$CMer08o/KELFlVrhL5QRO0 dn-webedit time-webedit transfer-system full-blind transfer-pattern 9.T ! ! ephone-dn 11 number 800 mwi on ! ! ephone-dn 12 number 801 mwi off ! ! line con 0 line aux 0 line 66 no activation-character no exec transport preferred none transport input all transport output pad telnet rlogin lapb-ta mop udptn v120 line vty 0 4 password cisco login ! scheduler allocate 20000 1000 end CME-SIP#
There is currently no verification procedure available for this configuration.
This section provides information you can use to troubleshoot your configuration.
A common cause for SIP IP Phones that are not able to get a dial tone is that there is another phone with the same extension. As of Cisco Unified Communications Manager Express 4.2, shared line is not supported on SIP Phones. Thus, SIP phones can not share the same extension among multiple phones. Additionally, make sure that the SIP phone is provisioned with a proper extension.
In order to resolve this issue, make sure that these occur:
The most likely causes for failure to be able to upgrade a phone is missing firmware files placed on the Cisco Unified Communications Manager Express flash or missing tftp-server commands.
Try these steps in order to resolve this issue:
In order to troubleshoot further, collect these debugs in order to see if the phone is able to get the appropriate phone loads from the Cisco Unified Communications Manager Express flash.
Debug tftp events
The most likely causes for not being able to provision is phone is that the phone does not have the proper IP address with TFTP server option.