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Table Of Contents
Configuring Phones to Make Basic Calls
Prerequisites for Configuring Phones to Make Basic Calls
Restrictions for Configuring Phones to Make Basic Calls
Information About Configuring Phones to Make Basic Calls
Digit Collection on SIP Phones
Session Transport Protocol for SIP Phones
How to Configure Phones for a PBX System
SCCP: Creating Directory Numbers
SCCP: Assigning Directory Numbers to Phones
SIP: Creating Directory Numbers
SIP: Assigning Directory Numbers to Phones
SIP: Verifying Dial Plan Configuration
SIP: Selecting Session-Transport Protocol for a Phone
SIP: Disabling SIP Proxy Registration for a Directory Number
Configuring Codec for Local Calling Between SIP and SCCP Phones
How to Configure Phones for a Key System
SCCP: Creating Directory Numbers for a Simple Key System
SCCP: Configuring Trunk Lines for a Key System
SCCP: Configuring Individual IP Phones for Key System
How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator
Troubleshooting Cisco ATA Support
Using Call Pickup and Group Call Pickup with Cisco ATA
SCCP: Configuring Analog Phone Support
SCCP: Verifying Analog Phone Support
SCCP: Configuring Cisco IP Communicator Support
SCCP: Verifying Cisco IP Communicator Support
SCCP: Troubleshooting Cisco IP Communicator Support
Configuration Examples for Making Basic Calls
Configuring SCCP Phones for Making Basic Calls: Example
Configuring SIP Phones for Making Basic Calls: Example
Disabling a Bulk Registration for a SIP Phone: Example
Remote Teleworker Phones: Example
Feature Information for Configuring Phones to Make Basic Calls
Configuring Phones to Make Basic Calls
Last Updated: September 25, 2007This module describes how to configure Cisco Unified IP phones in a Cisco Unified Communications Manager Express (Cisco Unified CME) system so that you can make and receive basic calls.
Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration, you can skip this module unless you want to modify the configuration to add phones.
Finding Feature Information in This Module
Your Cisco Unified CME version may not support all of the features documented in this module. For a list of the versions in which each feature is supported, see the "Feature Information for Configuring Phones to Make Basic Calls" section.
Contents
• Prerequisites for Configuring Phones to Make Basic Calls
• Restrictions for Configuring Phones to Make Basic Calls
• Information About Configuring Phones to Make Basic Calls
• How to Configure Phones for a PBX System
• How to Configure Phones for a Key System
• How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator
• Configuration Examples for Making Basic Calls
• Feature Information for Configuring Phones to Make Basic Calls
Prerequisites for Configuring Phones to Make Basic Calls
•Cisco IOS software and Cisco Unified CME software, including phone firmware files for Cisco Unified IP phones to be connected to Cisco Unified CME, must be installed in router flash memory. See "Installing and Upgrading Cisco Unified CME Software" on page 87.
•For Cisco Unified IP phones that are running SIP and are connected directly to Cisco Unified CME, Cisco Unified CME 3.4 or later must be installed on the router. See "Installing and Upgrading Cisco Unified CME Software" on page 87.
•Procedures in "Defining Network Parameters" on page 109 and "Configuring System-Level Parameters" on page 137 must be completed before you start the procedures in this section.
Restrictions for Configuring Phones to Make Basic Calls
•When you are configuring dial peers or ephone-dns, including park slots and conferencing extensions, on Cisco Integrated Services Router Voice Bundles, the following message may appear to warn you that free memory is not available:
%DIALPEER_DB-3-ADDPEER_MEM_THRESHOLD: Addition of dial-peers limited by available memory
To configure more dial peers or ephone-dns, increase the DRAM in the system. A moderately complex configuration may exceed the default 256 MB DRAM and require 512 MB DRAM. Note that many factors contribute to memory usage, in addition to the number of dial peers and ephone-dns configured.
Information About Configuring Phones to Make Basic Calls
To configure phones to make basic calls, you should understand the following concepts:
• Monitor Mode for Shared Lines
• Digit Collection on SIP Phones
• Session Transport Protocol for SIP Phones
Phones in Cisco Unified CME
An ephone, or "Ethernet phone," for SCCP or a voice-register pool for SIP is the software configuration for a phone in Cisco Unified CME. This phone can be either a Cisco Unified IP phone or an analog phone. Each physical phone in your system must be configured as an ephone or voice-register pool on the Cisco Unified CME router to receive support in the LAN environment. Each phone has a unique tag, or sequence number, to identify it during configuration.
Directory Numbers
A directory number, also known as an ephone-dn for SCCP or a voice-register dn for SIP, is the software configuration in Cisco Unified CME that represents the line connecting a voice channel to a phone. A directory number has one or more extension or telephone numbers associated with it to allow call connections to be made. Generally, a directory number is equivalent to a phone line, but not always. There are several types of directory numbers, which have different characteristics.
Each directory number has a unique dn-tag, or sequence number, to identify it during configuration. Directory numbers are assigned to line buttons on phones during configuration.
One virtual voice port and one or more dial peers are automatically created for each directory number, depending on the configuration for SCCP phones, or for SIP phones, when the phone registers in Cisco Unified CME.
The number of directory numbers that you create corresponds to the number of simultaneous calls that you can have, because each directory number represents a virtual voice port in the router. This means that if you want more than one call to the same number to be answered simultaneously, you need multiple directory numbers with the same destination number pattern.
The directory number is the basic building block of a Cisco Unified CME system. Six different types of directory number can be combined in different ways for different call coverage situations. Each type will help with a particular type of limitation or call-coverage need. For example, if you want to keep the number of directory numbers low and provide service to a large number of people, you might use shared directory numbers. Or if you have a limited quantity of extension numbers that you can use and you need to have a large quantity of simultaneous calls, you might create two or more directory numbers with the same number. The key is knowing how each type of directory number works and its advantages.
Not all types of directory numbers can be configured for all phones or for all protocols. In the remaining information about directory numbers, we have used SCCP in the examples presented but that does not imply exclusivity. The following sections describe the types of directory number in a Cisco Unified CME system:
• Two Directory Numbers with One Telephone or Extension Number
• Shared
• Monitor Mode for Shared Lines
• Overlaid
Single-Line
A single-line directory number has the following characteristics:
•Makes one call connection at a time using one phone line button. A single-line directory number has one telephone number associated with it.
•Should be used when phone buttons have a one-to-one correspondence to the PSTN lines that come into a Cisco Unified CME system.
•Should be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI), loopback, and music-on-hold (MOH) feed sources.
•When used with multiple-line features like call waiting, call transfer, and conferencing, there must be more than one single-line directory number on a phone.
•Can be combined with dual-line directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line or single-line when you initially create configuration entries. If you need to change from single-line to dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 7 shows a single-line directory number for an SCCP phone in Cisco Unified CME.
Figure 7 Single-Line Directory Number
Dual-Line
A dual-line directory number has the following characteristics:
•One voice port with two channels.
•Supported on IP phones that are running SCCP; not supported on IP phones that are running SIP.
•Can make two call connections at the same time using one phone line button. A dual-line directory number has two channels for separate call connections.
•Can have one number or two numbers (primary and secondary) associated with it.
•Should be used for a directory number that needs to use one line button for features like call waiting, call transfer, or conferencing.
•Cannot be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI), loopback, and music-on-hold (MOH) feed sources.
•Can be combined with single-line directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line or single-line when you initially create configuration entries. If you need to change from single-line to dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 8 shows a dual-line directory number for an SCCP phone in Cisco Unified CME.
Figure 8 Dual-Line Directory Number
Two Directory Numbers with One Telephone or Extension Number
Two directory numbers with one number have the following characteristics:
•Have the same telephone number but two separate virtual voice ports, and therefore can have two separate call connections.
•Can be dual-line (SCCP only) or single-line directory numbers.
•Can appear on the same phone on different buttons or on different phones.
•Should be used when you want the ability to make more call connections while using fewer numbers.
Figure 9 shows a phone with two buttons that have the same number, extension 1003. Each button has a different directory number (button 1 is directory number 13 and button 2 is directory number 14), so each button can make one independent call connection if the directory numbers are single-line and two call connections (for a total of four) if the directory numbers are dual-line.
Figure 10 shows two phones that each have a button with the same number. Because the buttons have different directory numbers, the calls that are connected on these buttons are independent of one another. The phone user at phone 4 can make a call on extension 1003, and the phone user on phone 5 can receive a different call on extension 1003 at the same time.
The two directory numbers-with-one-number situation is different than a shared line, which also has two buttons with one number but has only one directory number for both of them. A shared directory number will have the same call connection at all the buttons on which the shared directory number appears. If a call on a shared directory number is answered on one phone and then placed on hold, the call can be retrieved from the second phone on which the shared directory number appears. But when there are two directory numbers with one number, a call connection appears only on the phone and button at which the call is made or received. In the example in Figure 10, if the user at phone 4 makes a call on button 1 and puts it on hold, the call can be retrieved only from phone 4. For more information about shared lines, see the "Shared" section.
The examples in Figure 9 and Figure 10 show how two directory numbers with one number are used to provide a small hunt group capability. In Figure 9, if the directory number on button 1 is busy or does not answer, an incoming call to extension 1003 rolls over to the directory number associated with button 2 because the appropriate related commands are configured. Similarly, if button 1 on phone 4 is busy, an incoming call to 1003 rolls over to button 1 on phone 5.
Figure 9 Two Directory Numbers with One Number on One Phone
Figure 10 Two Directory Numbers with One Number on Two Phones
Dual-Number
A dual-number directory number has the following characteristics:
•Has two telephone numbers, a primary number and a secondary number.
•Can make one call connection if it is a single-line directory number.
•Can make two call connections at a time if it is a dual-line directory number (SCCP only).
•Should be used when you want to have two different numbers for the same button without using more than one directory number.
Figure 11 shows a directory number that has two numbers, extension 1006 and extension 1007.
Figure 11 Dual-Number Directory
Shared
A shared directory number has the following characteristics:
•Line appears on two different phones but uses the same directory number, and extension or phone number.
•Can make one call at a time and that call appears on both phones.
•Should be used when you want the capability to answer or pick up a call at more than one phone.
Because these phones share the same directory number, if the directory number is connected to a call on one phone, that directory number is unavailable for other calls on the second phone. If a call is placed on hold on one phone, it can be retrieved on the second phone. This is like having a single-line phone in your house with multiple extensions. You can answer the call from any phone on which the number appears, and you can pick it up from hold on any phone on which the number appears.
Figure 12 shows a shared directory number on phones that are running SCCP. Extension 1008 appears on both phone 7 and phone 8.
Figure 12 Shared Directory Number
Overlaid
An overlaid directory number has the following characteristics:
•Is a member of an overlay set, which includes all the directory numbers that have been assigned together to a particular phone button.
•Can have the same telephone or extension number as other members of the overlay set or different numbers.
•Can be single-line or dual-line, but cannot be mixed single-line and dual-line in the same overlay set.
•Can be shared on more than one phone.
Overlaid directory numbers provide call coverage similar to shared directory numbers because the same number can appear on more than one phone. The advantage of using two directory numbers in an overlay arrangement rather than as a simple shared line is that a call to the number on one phone does not block the use of the same number on the other phone, as would happen if it were a shared directory number.
For information about configuring call coverage using overlaid ephone-dns, see "Configuring Call-Coverage Features" on page 581.
You can overlay up to 25 lines on a single button. A typical use of overlaid directory numbers would be to create a "10x10" shared line, with ten lines in an overlay set shared by ten phones, resulting in the possibility of ten simultaneous calls to the same number. For configuration information, see the "SCCP: Creating Directory Numbers for a Simple Key System" section
Monitor Mode for Shared Lines
In Cisco CME 3.0 and later versions, Monitor mode for shared lines provides a visible line status indicating whether the line is in-use or not.
When the line is in use, it cannot be used for incoming or outgoing calls. A monitor-line lamp can be off or unlit only when its line is in the idle call state. The idle state occurs before a call is made and after a call is completed. For all other call states, the monitor line lamp is on or lit.
The line button for a monitored line can also be used as a direct-station-select for a call transfer when the monitored line is in an idle state. In this case, the receptionist who transfers a call from a normal line can press the Transfer button and then press the line button of the monitored line, causing the call to be transferred to the phone number of the monitored line.
For configuration information, see the "SCCP: Assigning Directory Numbers to Phones" section.
Monitor mode is intended for use only in the context of shared lines so that a receptionist can visually monitor the in-use status of several users' phone extensions; for example, for Busy Lamp Field (BLF) notification. To monitor all lines on an individual phone so that a receptionist can visually monitor the in-use status of that phone, see the "Watch Mode for Phones" section.
For BLF monitoring of speed-dial buttons and directory call-lists, see "Configuring Presence Service" on page 843.
Watch Mode for Phones
In Cisco Unified CME 4.1 and later versions, a line button that is configured for Watch mode on one phone provides Busy Lamp Field (BLF) notification for all lines on another phone (watched phone) for which the watched directory number is the primary line. Watch mode allows a phone user, such as a receptionist, to visually monitor the in-use status of an individual phone. The line and line button on the watching phone are available in watch mode for visual status only. Calls cannot be made or received using a line button that has been set in watch mode. Incoming calls on a line button that is in watch mode do not ring and do not display caller ID or call-waiting caller ID.
The line button for a watched phone can also be used as a direct-station-select for a call transfer when the watched phone is idle. In this case, the phone user who transfers a call from a normal line can press the Transfer button and then press the line button of the watched directory number, causing the call to be transferred to the phone number associated with the watched directory number.
For configuration information, see the "SCCP: Assigning Directory Numbers to Phones" section.
Note If the watched directory number is a shared line and the shared line is not idle on any phone with which it is associated, then in the context of watch mode, the status of the line button indicates that the watched phone is in use.
For best results when monitoring the status of an individual phone based on a watched directory number, the directory number configured for watch mode should not be a shared line. To monitor a shared line so that a receptionist can visually monitor the in-use status of several users' phone extensions, see the "Monitor Mode for Shared Lines" section.
For BLF monitoring of speed-dial buttons and directory call-lists, see "Configuring Presence Service" on page 843.
PSTN FXO Trunk Lines
In Cisco CME 3.2 and later, IP phones running SCCP can be configured to have buttons for dedicated PSTN FXO trunk lines, also known as FXO lines. FXO lines may used by companies whose employees require private PSTN numbers. For example, a salesperson may need a special number that customers can call without having to go through a main number. When a call comes in to the direct number, the salesperson knows that the caller is a customer. In the salesperson's absence the customer can leave voice mail. FXO lines can use PSTN service provider voice mail: when the line button is pressed, the line is seized, allowing the user to hear the stutter dial tone provided by the PSTN to indicate that voice messages are available.
Because FXO lines behave as private lines, users do not have to dial a prefix, such as 9 or 8, to reach an outside line. To reach phone users within the company, FXO-line users must dial numbers that use the company's PSTN number. For calls to nonPSTN destinations, such as local IP phones, a second directory number must be provisioned.
Calls placed to or received on an FXO line have restricted Cisco Unified CME services and cannot be transferred by Cisco Unified CME. However, phone users are able to access hookflash-controlled PSTN services using the Flash soft key.
In Cisco Unified CME 4.0, the following FXO trunk enhancements were introduced to improve the keyswitch emulation behavior of PSTN lines on phones running SCCP, in a Cisco Unified CME system.
•FXO port monitoring—Allows the line button on IP phones to reliably show the status of an FXO port when the port is in use. The status indicator, either a lamp or an icon, depending on the phone model, accurately displays the status of the FXO port during the duration of the call, even after the call is forwarded or transferred. The same FXO port can be monitored by multiple phones using multiple trunk ephone-dns.
•Transfer recall—If a transfer-to phone does not answer after a specified timeout, the call is returned to the phone that initiated the transfer and it resumes ringing on the FXO line button. The directory number must be dual-lined.
•Transfer-to button optimization—When an FXO call is transferred to a private extension button on another phone, and that phone has a shared line button for the FXO port, after the transfer is committed and the call is answered, the connected call displays on the FXO line button of the transfer-to phone. This frees up the private extension line on the transfer-to phone. The directory number n must be dual-line.
•Dual-line ephone-dns— Directory numbers for FXO lines can now be configured for dual-line to support the FXO monitoring, transfer recall, and transfer-to button optimization features.
For configuration information, see the "SCCP: Configuring Trunk Lines for a Key System" section.
Analog Phones
Cisco Unified CME supports analog phones using Cisco Analog Telephone Adaptors (ATAs) or FXS ports in SCCP mode or H.323 mode, and supports fax machines on Cisco ATA or FXS ports in H.323 mode. The FXS ports used for analog phones or fax can be on the Cisco Unified CME router or on a Cisco VG 224 voice gateway or Integrated Services Router (ISR). This section provides information on the following topics:
Cisco ATAs in SCCP Mode
You can configure the Cisco ATA 186 or Cisco ATA 188 to cost-effectively support analog phones using SCCP in Cisco IOS Release 12.2(11)T and later. Each Cisco ATA enables two analog phones to function as IP phones. For configuration information, see the "Configuring Cisco ATA Support" section.
FXS Ports in SCCP Mode
FXS ports on Cisco VG 224 Voice Gateways and Cisco 2800 Series and Cisco 3800 Series ISRs can be configured for SCCP supplementary features. For information about using SCCP supplementary features on analog FXS ports on a Cisco IOS gateway under the control of a Cisco Unified CME router, see SCCP Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways.
FXS Ports in H.323 Mode
FXS ports on platforms that cannot enable SCCP supplementary features can use H.323 mode to support call waiting, caller ID, hookflash transfer, modem pass-through, fax (T.38, Cisco fax relay, and pass-through), and PLAR. These features are provisioned as Cisco IOS voice features and not as Cisco Unified CME features. Note that when using Cisco Unified CME, you can configure FXS ports in H.323 mode for call waiting or hookflash transfer, but not both at the same time.
The following links provide details on configuring analog phone features for FXS ports in H.323 mode:
• "Configuring Analog Voice Ports" section in Voice Ports Configuration Guide
•" Caller ID" section of the Cisco IOS Voice Configuration Library for your Cisco IOS release
•" Modem Support for VoIP" section of the Cisco IOS Voice Configuration Library for your Cisco IOS release
• Cisco IOS Fax and Modem Services over IP Application Guide for your Cisco IOS release
Remote Teleworker Phones
IP phones or instances of Cisco IP Communicator can be connected to a Cisco Unified CME system over a wide area network (WAN) to support teleworkers who have offices that are remote from the Cisco Unified CME router. The maximum number of remote phones that can be supported is determined by the available bandwidth.
IP addressing is a determining factor in the most critical aspect of remote teleworker phone design. The following two scenarios represent the most common designs, the second one is the most common for small and medium businesses:
•Remote site IP phones and the hub Cisco Unified CME router use globally routable IP addresses.
•Remote site IP phones use NAT with nonroutable private IP addresses and the hub Cisco Unified CME router uses a globally routable address (see Figure 13). This scenario results in one-way audio unless you use one of the following workarounds:
–Configure static NAT mapping on the remote site router (for example, a Cisco 831 Ethernet Broadband Router) to convert between a private address and a globally routable address. This solution uses fewer Cisco Unified CME resources, but voice is unencryped across the WAN.
–Configure an IPsec VPN tunnel between the remote site router (or example, a Cisco 831) and the Cisco Unified CME router. This solution requires an Advanced IP Services or higher image on the Cisco Unified CME router if this router is used to terminate the VPN tunnel. Voice will be encrypted across the WAN. This method will also work with the Cisco VPN client on a PC to support Cisco IP Communicator.
Figure 13 Remote Site IP Phones Using NAT
Media Termination Point for Remote Phones
Media termination point (MTP) configuration is used to ensure that Real-Time Transport Protocol (RTP) media packets from remote phones always transit through the Cisco Unified CME router. Without the MTP feature, a phone that is connected in a call with another phone in the same Cisco Unified CME system sends its media packets directly to the other phone, without the packets going through the Cisco Unified CME router. MTP forces the packets to be sourced from the Cisco Unified CME router.
When this configuration is used to instruct a phone to always send its media packets to the Cisco Unified CME router, the router acts as an MTP or proxy and forwards the packets to the destination phone. If a firewall is present, it can be configured to pass the RTP packets because the router uses a specified UDP port for media packets. In this way, RTP packets from remote IP phones can be delivered to IP phones on the same system though they must pass through a firewall.
You must use the mtp command to explicitly enable MTP for each remote phone that sends media packets to Cisco Unified CME.
One factor to consider is whether you are using multicast music on hold (MOH) in your system. Multicast packets generally cannot be forwarded to phones that are reached over a WAN. The multicast MOH feature checks to see if MTP is enabled for a phone and if it is, MOH is not sent to that phone. If you have a WAN configuration that can forward multicast packets and you can allow RTP packets through your firewall, you can decide not to use MTP.
For configuration information, see the "SCCP: Enabling a Remote Phone" section.
G.729r8 Codec on Remote Phones
You can select the G.729r8 codec on a remote IP phone to help save network bandwidth. The default codec is G.711 mu-law. If you use the codec g729r8 command without the dspfarm-assist keyword, the use of the G.729 codec is preserved only for calls between two phones on the Cisco Unified CME router (such as between an IP phone and another IP phone or between an IP phone and an FXS analog phone). The codec g729r8 command has no affect on a call directed through a VoIP dial peer unless the dspfarm-assist keyword is also used.
For configuration information, see the "SCCP: Enabling a Remote Phone" section.
For information about transcoding behavior when using the G.729r8 codec, see the "Transcoding When a Remote Phone Uses G.729r8" section on page 325.
Digit Collection on SIP Phones
Digit strings dialed by phone users must be collected and matched against predefined patterns to place calls to the destination corresponding to the user's input. Before Cisco Unified CME 4.1, SIP phone users had to press the DIAL soft key or # key, or wait for the interdigit-timeout to trigger call processing. In Cisco United CME 4.1 and later, two methods of collecting and matching digits are supported for SIP phones, depending on the model of phone:
KPML Digit Collection
Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user input digit by digit. Each digit dialed by the phone user generates its own signaling message to Cisco Unified CME, which performs pattern recognition by matching a destination pattern to a dial peer as it collects the dialed digits. This process of relaying each digit immediately is similar to the process used by SCCP phones. It eliminates the need for the user to press the Dial soft key or wait for the interdigit timeout before the digits are sent to Cisco Unified CME for processing.
KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For configuration information, see the "SIP: Enabling KPML" section.
SIP Dial Plans
A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete after a user goes off-hook and dials a destination number. Dial plans allow SIP phones to perform local digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the SIP phone sends an INVITE message to Cisco Unified CME to initiate the call to the number matching the user's input. All of the digits entered by the user are presented as a block to Cisco Unified CME for processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead compared to KPML digit collection.
SIP dial plans eliminate the need for a user to press the Dial soft key or # key, or to wait for the interdigit timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a SIP phone. The dial plan is downloaded to the phone in the configuration file.
You can configure SIP dial plans and associate them with the following SIP phones:
•Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE—These phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan has priority.
If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit timeout before the SIP NOTIFY message is sent to Cisco Unified CME. Unlike other SIP phones, these phones do not have a Dial soft key to indicate the end of dialing, except when on-hook dialing is used. In this case, the user can press the Dial soft key at any time to send all the dialed digits to Cisco Unified CME.
•Cisco Unified IP Phone 7905, 7912, 7940, and 7960—These phones use dial plans and do not support KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do not match a dial plan, the user must press the Dial soft key or wait for the interdigit timeout before digits are sent to Cisco Unified CME.
When you reset a phone, the phone requests its configuration files from the TFTP server, which builds the appropriate configuration files depending on the type of phone.
•Cisco Unified IP Phone 7905 and 7912—The dial plan is a field in their configuration files.
•Cisco Unified IP Phone 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and 7971GE—The dial plan is a separate XML file that is pointed to from the normal configuration file.
For configuration information for Cisco Unified CME, see the "SIP: Configuring Dial Plans" section.
Session Transport Protocol for SIP Phones
In Cisco Unified CME 4.1 and later versions, you can select TCP as the transport protocol for connecting supported SIP phones to Cisco Unified CME. Previously only UDP was supported. TCP is selected for individual SIP phones by using the session-transport command in voice register pool or voice register template configuration mode. For configuration information, see the "SIP: Selecting Session-Transport Protocol for a Phone" section.
How to Configure Phones for a PBX System
This section contains the following tasks:
• SCCP: Creating Directory Numbers (required)
• SCCP: Assigning Directory Numbers to Phones (required)
• SIP: Creating Directory Numbers (required)
• SIP: Assigning Directory Numbers to Phones (required)
• SIP: Configuring Dial Plans (optional)
• SIP: Verifying Dial Plan Configuration (optional)
• SIP: Enabling KPML (optional)
• SIP: Selecting Session-Transport Protocol for a Phone (optional)
• SIP: Disabling SIP Proxy Registration for a Directory Number (required)
• Configuring Codec for Local Calling Between SIP and SCCP Phones (required)
SCCP: Creating Directory Numbers
To create a directory number in Cisco Unified CME for a SCCP phone, intercom line, voice port, or a message-waiting indicator (MWI), perform the following steps for each directory number to be created. Each ephone-dn becomes a virtual line, or extension, on which call connections can be made. Each ephone-dn configuration automatically creates one or more virtual dial peers and virtual voice ports to make those call connections.
Note To create and assign directory numbers to be included in an overlay set, see "SCCP: Configuring Overlaid Ephone-dns" on page 633.
Prerequisites
•The maximum number of directory numbers must be configured for other than the default, by using the max-dn command.
Restrictions
•The Cisco Unified IP Phone 7931G is a SCCP keyset phone and when configured for a key system, does not support the dual-line option for a directory number. To configure a Cisco Unified IP Phone 7931G, see the "How to Configure Phones for a Key System" section.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. name name
6. end
DETAILED STEPS
What to Do Next
After creating directory numbers, you can assign one or more directory number to a Cisco Unified IP phone. See "SCCP: Assigning Directory Numbers to Phones" section.
SCCP: Assigning Directory Numbers to Phones
This task sets up the initial ephone-dn-to-ephone relationships—that is, how and which extensions appear on each phone. To create and modify phone-specific parameters for individual SCCP phones, perform the following steps for each SCCP phone to be connected in Cisco Unified CME.
Note To create and assign directory numbers to be included in an overlay set, see "SCCP: Configuring Overlaid Ephone-dns" on page 633.
Prerequisites
•To configure a phone line for Watch (w) mode by using the button command, Cisco Unified CME 4.1 or a later version.
•To configure a phone line for Monitor (m) mode by using the button command, Cisco CME 3.0 or a later version.
Restrictions
•For Watch mode. If the watched directory number is associated with several phones, then the watched phone is the one on which the watched directory number is on button 1 or the one on which the watched directory number is on the button that is configured by using the auto-line command, with auto-line having priority. For configuration information, see "Configuring Automatic Line Selection" on page 479
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type [addon 1 module-type [2 module-type]]
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number] [button-number...]
7. keypad-normalize
8. end
DETAILED STEPS
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267.
Examples
The following example assigns extension 2225 in the Accounting Department to button 1 on ephone 2.
ephone-dn 25
number 2225
name Accounting
ephone 2
mac-address 00E1.CB13.0395
type 7960
button 1:25
SIP: Creating Directory Numbers
To create a directory number in Cisco Unified CME for a SIP phone, intercom line, voice port, or a message-waiting indicator (MWI), perform the following steps for each directory number to be created. Each directory number becomes a virtual line, or extension, on which call connections can be made. Each directory number configuration automatically creates one or more virtual dial peers and virtual voice ports to make those call connections.
Prerequisites
•Cisco CME 3.4 or a later version.
•The maximum number of directory numbers supported by a router is version and platform dependent. To configure more directory numbers than the default, use the max-dn (voice register global) command before performing this procedure. For configuration information, see "SIP: Setting Up Cisco Unified CME" on page 153.
Restrictions
•Call forward all, presence, and message-waiting indication (MWI) features in Cisco Unified CME 4.1 and later versions require that SIP phones are configured with a directory number (using dn keyword in number command); direct line numbers are not supported.
•SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only.
•Shared lines are not supported by SIP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. end
DETAILED STEPS
SIP: Assigning Directory Numbers to Phones
This task sets up which extensions appear on each phone. To create and modify phone-specific parameters for individual SIP phones, perform the following steps for each SIP phone to be connected in Cisco Unified CME.
Note If your Cisco Unified CME system supports SCCP and also SIP phones, do not connect your SIP phones to your network until after you have verified the configuration profile for the SIP phone.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. id mac address
5. type phone-type
6. number tag dn dn-tag
7. username name password string
8. dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
9. end
DETAILED STEPS
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•If you want to select the session-transport protocol for a SIP phone, see the "SIP: Selecting Session-Transport Protocol for a Phone" section.
•If you are finished configuring phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SIP: Generating Configuration Profiles for SIP Phones" on page 270.
SIP: Configuring Dial Plans
Dial plans enable SIP phones to recognize digit strings dialed by users. After the phone recognizes a dial pattern, it automatically sends a SIP INVITE message to Cisco Unified CME to initiate the call and does not require the user to press the Dial key or wait for the interdigit timeout. To define a dial plan for a SIP phone, perform the following steps.
Prerequisites
•Cisco Unified CME 4.1 or a later version.
•mode cme command must be enabled in Cisco Unified CME.
Restrictions
•If you create a dial plan by downloading a custom XML dial pattern file to flash and using the filename command, and the XML file contains an error, the dial plan might not work properly on a phone. We recommend creating a dial pattern file using the pattern command.
•To remove a dial plan that was created using a custom XML file with the filename command, you must remove the dial plan from the phone, create a new configuration profile, and then use the reset command to reboot the phone. You can use the restart command after removing a dial plan from a phone only if the dial plan was created using the pattern command.
•To use KPML if a matching dial plan is not found, when both a dial plan and KPML are enabled on a phone, you must configure a dial pattern with a single wildcard character (.) as the last pattern in the dial plan. For example:
voice register dialplan 10
type 7940-7960-others
pattern 1 66...
pattern 2 91.......
pattern 3 .
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dialplan dialplan-tag
4. type phone-type
5. pattern tag string [button button-number] [timeout seconds] [user {ip | phone}]
or
filename filename6. exit
7. voice register pool pool-tag
8. dialplan dialplan-tag
9. end
DETAILED STEPS
Command or Action PurposeStep 1
enable
Example:Router> enable
Enables privileged EXEC mode.
•Enter your password if prompted.
Step 2
configure terminal
Example:Router# configure terminal
Enters global configuration mode.
Step 3
voice register dialplan dialplan-tag
Example:Router(config)# voice register dialplan 1
Enters voice register dialplan configuration mode to define a dial plan for SIP phones.
Step 4
type phone-type
Example:Router(config-register-dialplan)# type 7905-7912
Defines a phone type for the SIP dial plan.
•7905-7912—Cisco Unified IP Phone 7905, 7905G, 7912, or 7912G.
•7940-7960-others—Cisco Unified IP Phone 7911, 7940, 7940G, 7941, 7941GE, 7960, 7960G, 7961, 7961GE, 7970, or 7971.
•The phone type specified with this command must match the type of phone for which the dial plan is used. If this phone type does not match the type assigned to the phone with the type command in voice register pool mode, the dial-plan configuration file is not generated.
•You must enter this command before using the pattern or filename command in the next step.
Step 5
pattern tag string [button button-number] [timeout seconds] [user {ip | phone}]
or
filename filename
Example:Router(config-register-dialplan)# pattern 1 52...
or
Router(config-register-dialplan)# filename dialsip
Defines a dial pattern for a SIP dial plan.
•tag—Number that identifies the dial pattern. Range: 1 to 24.
•string—Dial pattern, such as the area code, prefix, and first one or two digits of the telephone number, plus wildcard characters or dots (.) for the remainder of the dialed digits.
•button button-number—(Optional) Button to which the dial pattern applies.
•timeout seconds—(Optional) Time, in seconds, that the system waits before dialing the number entered by the user. Range: 0 to 30. To have the number dialed immediately, specify 0. If you do not use this parameter, the phone's default interdigit timeout value is used (10 seconds).
•user—(Optional) Tag that automatically gets added to the dialed number. Do not use this keyword if Cisco Unified CME is the only SIP call agent.
•ip—Uses the IP address of the user.
•phone—Uses the phone number of the user.
•Repeat this command for each pattern that you want to include in this dial plan.
or
Specifies a custom XML file that contains the dial patterns to use for the SIP dial plan.
•You must load the custom XML file must into flash and the filename cannot include the .xml extension.
•The filename command is not supported for the Cisco Unified IP Phone 7905 or 7912.
Step 6
exit
Example:Router(config-register-dialplan)# exit
Exits dialplan configuration mode.
Step 7
voice register pool pool-tag
Example:Router(config)# voice register pool 4
Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.
•pool-tag—Unique sequence number of the SIP phone to be configured. Range is version and platform-dependent; type ? to display range. You can modify the upper limit for this argument with the max-pool command.
Step 8
dialplan dialplan-tag
Example:Router(config-register-pool)# dialplan 1
Assigns a dial plan to a SIP phone.
•dialplan-tag—Number that identifies the dial plan to use for this SIP phone. This is the number that was used with the voice register dialplan command in Step 3. Range: 1 to 24.
Step 9
end
Example:Router(config-register-global)# end
Exits to privileged EXEC mode.
Examples
The following example shows the configuration for dial plan 1 which is assigned to SIP phone 1.
voice register dialplan 1
type 7940-7960-others
pattern 1 2... timeout 10 user ip
pattern 2 1234 user ip button 4
pattern 3 65...
pattern 4 1...!
!
voice register pool 1
id mac 0016.9DEF.1A70
type 7961GE
number 1 dn 1
number 2 dn 2
dialplan 1
dtmf-relay rtp-nte
codec g711ulaw
What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and restart the phones. See "Generating Configuration Files for Phones" on page 265.
SIP: Verifying Dial Plan Configuration
Step 1 show voice register dialplan tag
This command displays the configuration information for a specific SIP dial plan.
Router# show voice register dialplan 1
Dialplan Tag 1
Config:
Type is 7940-7960-others
Pattern 1 is 2..., timeout is 10, user option is ip, button is default
Pattern 2 is 1234, timeout is 0, user option is ip, button is 4
Pattern 3 is 65..., timeout is 0, user option is phone, button is default
Pattern 4 is 1..., timeout is 0, user option is phone, button is default
Step 2 show voice register pool tag
This command displays the dial plan assigned to a specific SIP phone.
Router# show voice register pool 29
Pool Tag 29
Config:
Mac address is 0012.7F54.EDC6
Number list 1 : DN 29
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
keep-conference is enabled
dialplan tag is 1
kpml signal is enabled
service-control mechanism is not supported
.
.
.
Step 3 show voice register template tag
This command displays the dial plan assigned to a specific template.
Router# show voice register template 3
Temp Tag 3
Config:
Attended Transfer is disabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Conference is enabled
Caller-ID block is disabled
DnD control is enabled
Anonymous call block is disabled
Voicemail is 62000, timeout 15
Dialplan Tag is 1
Transport type is tcp
SIP: Enabling KPML
To enable KPML digit collection on a SIP phone, perform the following steps.
Prerequisites
•Cisco Unified CME 4.1 or a later version.
Restrictions
•This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
•A dial plan assigned to a phone has priority over KPML.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. digit collect kpml
5. end
6. show voice register dial-peer
DETAILED STEPS
What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and restart the phones. See "Generating Configuration Files for Phones" on page 265.
SIP: Selecting Session-Transport Protocol for a Phone
To change the session-transport protocol for a SIP phone to TCP, from the default of UDP, perform the following steps.
Prerequisites
•Cisco Unified CME 4.1 or a later version.
•SIP phone to which configuration is to be applied must be already configured. For configuration information, see the "SIP: Assigning Directory Numbers to Phones" section.
Restrictions
•TCP is not supported as a session-transport protocol for the Cisco Unified IP Phone 7905, 7912, 7940, or 7960. If TCP is assigned to an unsupported phone using this command, calls to that phone will not complete successfully. The phone can originate calls, but it uses UDP, although TCP has been assigned.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. session-transport {tcp | udp}
5. end
DETAILED STEPS
What to Do Next
•If you want to disable SIP Proxy registration for an individual directory number, see the "SIP: Disabling SIP Proxy Registration for a Directory Number" section.
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•If you are finished configuring phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SIP: Generating Configuration Profiles for SIP Phones" on page 270
SIP: Disabling SIP Proxy Registration for a Directory Number
To prevent a particular directory number from registering with an external SIP proxy server, perform the following steps.
Prerequisites
•Cisco Unified CME 3.4 or a later version.
•Bulk registration is configured at system level. For configuration information, see "Configuring Bulk Registration" on page 141.
Restrictions
•Phone numbers that are registered under voice register dn must belong to a SIP phone that is itself registered in Cisco Unified CME.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. no-reg
6. end
DETAILED STEPS
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•If you are finished configuring phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SIP: Generating Configuration Profiles for SIP Phones" on page 270
Configuring Codec for Local Calling Between SIP and SCCP Phones
To designate a codec for individual phones to ensure connectivity between SIP and SCCP phones connected to the same Cisco Unified CME router, perform the following steps for each SIP or SCCP phone.
Note If codec values for the dial peers of an internal connection do not match, the call fails.
Prerequisites
•Cisco Unified CME 3.4 or a later version.
•Cisco Unified IP phone to which codec is to be applied must be already configured. For configuration information for SIP phones, see the "SIP: Assigning Directory Numbers to Phones" section. For configuration information for SCCP phones, see the "SCCP: Assigning Directory Numbers to Phones" section.
Restrictions
•Required only if you have SIP and SCCP phones connected to the same Cisco Unified CME router.
•Modify the configuration for either SIP or SCCP phones to ensure that the codec for all phones match. Do not modify the configuration for SIP and SCCP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone ephone-tag
or
voice register pool-tag4. codec codec-type
5. end
DETAILED STEPS
What to Do Next
•If you want to select the session-transport protocol for a SIP phone, see the "SIP: Selecting Session-Transport Protocol for a Phone" section.
•If you are finished configuring SIP phones to make basic calls using, you are ready to generate configuration files for the phones to be connected. See "SIP: Generating Configuration Profiles for SIP Phones" on page 270.
•If you are finished configuring SCCP phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267.
How to Configure Phones for a Key System
This section contains the following tasks:
• SCCP: Creating Directory Numbers for a Simple Key System (required)
• SCCP: Configuring Trunk Lines for a Key System (required)
• SCCP: Configuring Individual IP Phones for Key System (required)
SCCP: Creating Directory Numbers for a Simple Key System
To create a set of directory numbers with the same number to be associated with multiple line buttons on an IP phone and provide support for call waiting and call transfer on a key system phone, perform the following steps.
Restrictions
•Do not configure directory numbers for a key system for dual-line mode because this does not conform to the key system one-call-per-line button usage model for which the phone is designed.
•Provisioning support for the Cisco Unified IP Phone 7931 is available only in Cisco Unified CME 4.0(2) and later versions.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. preference preference-order
6. no huntstop
or
huntstop7. mwi-type {visual | audio | both}
8. end
DETAILED STEPS
Examples
The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:10 2:11 3:12 4:13 5:14 6:15
SCCP: Configuring Trunk Lines for a Key System
To set up trunk lines for your key system, perform only one of the following procedures:
•To only enable direct status monitoring of the FXO port on the line button of the IP phone, see the "SCCP: Configuring a Simple Key System Phone Trunk Line Configuration" section
•To enable direct status monitoring and allow transferred PSTN FXO line calls to be automatically recalled if the transfer target does not answer, see the "SCCP: Configuring an Advanced Key System Phone Trunk Line Configuration" section.
SCCP: Configuring a Simple Key System Phone Trunk Line Configuration
Perform the steps in this section to:
•Create directory numbers corresponding to each FXO line that allows phones to have shared or private lines connected directly to the PSTN.
•Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port during the duration of the call.
Prerequisites
•FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number
•Dial peers for FXO port must be configured; for example:
dial-peer voice 111 pots
destination-pattern 811 <<----Trunk-tag
port 1/0/0
Restrictions
•A directory number with a trunk line cannot be configured for call forward, busy, or no answer.
•Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed on IP phones.
•Numbers entered after trunk line is seized will not appear in call history or call detail records (CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.
•FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and NewCall soft keys.
•FXO trunk lines do not support conference initiator dropoff.
•FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk line before pressing the Redial button.
•FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP phone connected. The conference initiator is unable to participate in the conference, but can place calls on other lines.
•FXO trunk lines do not support bulk speed dial.
•FXO port monitoring has the following restrictions:
–Not supported before Cisco Unified CME 4.0.
–Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports. FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
–Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
–T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot into a ds0-group).
•Transfer recall and transfer-to button optimization are supported on dual-line directory numbers only in Cisco Unified CME 4.0 and later.
•Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on hold, or call pickup at alert.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] monitor-port port
6. end
DETAILED STEPS
Examples
The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone-dn 51
number 801
trunk 811 monitor-port 1/0/0
ephone-dn 52
number 802
trunk 812 monitor-port 1/0/1
ephone-dn 53
number 803
trunk 813 monitor-port 1/0/2
ephone-dn 54
number 804
trunk 814 monitor-port 1/0/3
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54
voice-port 1/0/0
connection plar opx 801
voice-port 1/0/1
connection plar opx 802
voice-port 1/0/2
connection plar opx 803
voice-port 1/0/3
connection plar opx 804
dial-peer voice 811 pots
destination-pattern 811
port 1/0/0
dial-peer voice 812 pots
destination-pattern 812
port 1/0/1
dial-peer voice 813 pots
destination-pattern 813
port 1/0/2
dial-peer voice 814 pots
destination-pattern 814
port 1/0/3
What to Do Next
You are ready to configure each individual phone and assign button numbers, line characteristics, and directory numbers to buttons on the phone. See the "SCCP: Configuring Individual IP Phones for Key System" section.
SCCP: Configuring an Advanced Key System Phone Trunk Line Configuration
Perform the steps in this section to:
•Create directory numbers corresponding to each FXO line that allows phones to have shared or private lines connected directly to the PSTN.
•Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port during the duration of the call.
•Allow transferred PSTN FXO line calls to be automatically recalled if the transfer target does not answer after the specified number of seconds. The call is withdrawn from the transfer-to phone and the call resumes ringing on the phone that initiated the transfer.
Prerequisites
•FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number
•Dial peers for FXO port must be configured; for example:
dial-peer voice 111 pots
destination-pattern 811 <<----Trunk-tag
port 1/0/0
Restrictions
•An ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.
•Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed on IP phones.
•Numbers entered after trunk line is seized will not appear in call history or call detail records (CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.
•FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and NewCall soft keys.
•FXO trunk lines do not support conference initiator dropoff.
•FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk line before pressing the Redial button.
•FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP phone connected. The conference initiator is unable to participate in the conference, but can place calls on other lines.
•FXO trunk lines do not support bulk speed dial.
•FXO port monitoring has the following restrictions:
–Not supported before Cisco Unified CME 4.0.
–Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports. FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
–Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
–T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot into a ds0-group).
•Transfer recall and transfer-to button optimization is supported on dual-line directory numbers only in Cisco Unified CME 4.0 and later.
•Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on hold, or call pickup at alert.
•Transfer recall is not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag dual-line
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] [transfer-timeout seconds] [monitor-port port]
6. huntstop [channel]
7. end
DETAILED STEPS
Examples
The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10. These four PSTN line appearances are configured as dual lines to provide a second call channel on which to place an outbound consultation call during a call transfer attempt. This configuration allows the phone to remain part of the call in order to monitor the progress of the transfer attempt, and if the transfer is not answered, to pull the call back to the phone on the original PSTN line button.
ephone-dn 10
number 101
no huntstop
ephone-dn 11
number 101
preference 1
no huntstop
ephone-dn 12
number 101
preference 2
no huntstop
ephone-dn 13
number 101
preference 3
no huntstop
ephone-dn 14
number 101
preference 4
no huntstop
ephone-dn 15
number 101
preference 5
ephone-dn 51 dual-line
number 801
trunk 811 transfer-timeout 30 monitor-port 1/0/0
huntstop channel
ephone-dn 52 dual-line
number 802
trunk 812 transfer-timeout 30 monitor-port 1/0/1
huntstop channel
ephone-dn 53 dual-line
number 803
trunk 813 transfer-timeout 30 monitor-port 1/0/2
huntstop channel
ephone-dn 54 dual-line
number 804
trunk 814 transfer-timeout 30 monitor-port 1/0/3
huntstop channel
ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54
voice-port 1/0/0
connection plar opx 801
voice-port 1/0/1
connection plar opx 802
voice-port 1/0/2
connection plar opx 803
voice-port 1/0/3
connection plar opx 804
dial-peer voice 811 pots
destination-pattern 811
port 1/0/0
dial-peer voice 812 pots
destination-pattern 812
port 1/0/1
dial-peer voice 813 pots
destination-pattern 813
port 1/0/2
dial-peer voice 814 pots
destination-pattern 814
port 1/0/3
SCCP: Configuring Individual IP Phones for Key System
To assign button numbers, line characteristics, and directory numbers to buttons on an individual phone to operate as a key system phone, perform the following steps.
Restrictions
•Provisioning for Cisco Unified IP Phone 7931G is available only in Cisco Unified CME 4.0(2) and later versions.
•Cisco Unified IP Phone 7931G can support only one call waiting overlaid per directory number.
•Cisco Unified IP Phone 7931G cannot support overlays that contain directory numbers configured for dual-line mode.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number] [button-number...]
7. mwi-line line-number
8. end
DETAILED STEPS
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see "SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G" on page 943.
•If you are finished configuring phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267.
How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator
This section contains the following tasks:
• Configuring Cisco ATA Support (required)
• Verifying Cisco ATA Support (optional)
• Using Call Pickup and Group Call Pickup with Cisco ATA (optional)
• SCCP: Configuring Analog Phone Support (required)
• SCCP: Verifying Analog Phone Support (optional)
• SCCP: Enabling a Remote Phone (required)
• SCCP: Verifying Remote Phones (optional)
• SCCP: Configuring Cisco IP Communicator Support (required)
• SCCP: Troubleshooting Cisco IP Communicator Support (optional)
Configuring Cisco ATA Support
To enable an analog phone that uses a Cisco ATA to register with Cisco Unified CME, perform the following steps.
Restrictions
For a Cisco ATA that is registered to a Cisco Unified CME system to participate in fax calls, it must have its ConnectMode parameter set to use the same RTP payload type as the Cisco voice gateway that is performing the fax pass-through. Cisco voice gateways use standard payload type 0/8, which is selected on Cisco ATAs by setting bit 2 of the ConnectMode parameter to 1. For more information, see the " Parameters and Defaults" chapter in the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).
SUMMARY STEPS
1. Install Cisco ATA.
2. Configure Cisco ATA for SCCP.
3. Upgrade firmware.
4. Set network parameters on Cisco ATA.
5. Configure analog phones in Cisco Unified CME.
DETAILED STEPS
Step 1 Install the Cisco ATA. See the "Installing the Cisco ATA" chapter in the in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).
Step 2 Configure the Cisco ATA. See the "Configuring the Cisco ATA for SCCP" chapter in the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).
Step 3 Upgrade to the latest Cisco ATA image. If you are using either the v2.14 or v2.14ms Cisco ATA 186 image based on the 2.14 020315a build for H.323/SIP or the 2.14 020415a build for MGCP or SCCP, you must upgrade to the latest version to install a security patch. This patch fixes a security hole in the Cisco ATA Web server that allows users to bypass the user interface password.
For information about upgrading firmware, see "Installing and Upgrading Cisco Unified CME Software" on page 87. Alternatively, you can use a manual method, as described in the "Upgrading the Cisco ATA Signaling Image" chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).
Step 4 Configure the Cisco ATA to set the following parameters:
–DHCP parameter to 1 (enabled).
–TFTP parameter to 1 (enabled).
–TFTPURL parameter to the IP address of the router running Cisco Unified CME.
–SID0 parameter to a period (.) or the MAC address of the Cisco ATA (to enable the first port).
–SID1 parameter to a period (.) or a modified version the Cisco ATA's MAC address, with the first two hexadecimal numbers removed and 01 appended to the end, if you want to use the second port. For example, if the MAC address of the Cisco ATA is 00012D01073D, set SID1 to 012D01073D01.
–Nprintf parameter to the IP address and port number of the host to which all Cisco ATA debug messages are sent. The port number is usually set to 9001.
–To prevent tampering and unauthorized access to the Cisco ATA 186, you can disable the web-based configuration. However, if you disable the web configuration page, you must use either a TFTP server or the voice configuration menu to configure the Cisco ATA 186.
Step 5 Configure analog phones that use a Cisco ATA in the same way as a Cisco Unified IP phone. In the type command, use the ata keyword. For information on how to provision phones, see the "SCCP: Creating Directory Numbers" section.
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see "SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G" on page 943.
•If you are finished configuring phones to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267 and "SIP: Generating Configuration Profiles for SIP Phones" on page 270.
Verifying Cisco ATA Support
Use the show ephone ata command to display SCCP phone configurations with the type ata command.
The following is sample output for a Cisco Unified CME configured for two analog phones using a Cisco ATA with MAC address 000F.F758.E70E.
ephone-30 Mac:000F.F758.E70E TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 1 and Server in ver 1
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:1.4.188.72 15325 ATA Phone keepalive 7 max_line 2 dual-line
button 1: dn 80 number 8080 CH1 IDLE CH2 IDLE
ephone-31 Mac:0FF7.58E7.0E01 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 1 and Server in ver 1
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:3
IP:1.4.188.72 15400 ATA Phone keepalive 7 max_line 2 dual-line
button 1: dn 81 number 8081 CH1 IDLE CH2 IDLE
Troubleshooting Cisco ATA Support
Use the debug ephone detail command to diagnose problems with analog phones that use Cisco ATAs. For more information, see the Cisco IOS Debug Command Reference for your Cisco IOS release.
The following is sample output for two analog phones using a Cisco ATA with MAC address 000F.F758.E70E. The sample shows the activities that take place when the phones register.
Router# debug ephone detail mac-address 000F.F758.E70E
*Apr 5 02:50:11.966: New Skinny socket accepted [1] (33 active)
*Apr 5 02:50:11.970: sin_family 2, sin_port 15325, in_addr 1.4.188.72
*Apr 5 02:50:11.970: skinny_add_socket 1 1.4.188.72 15325
21:21:49: %IPPHONE-6-REG_ALARM: Name=ATA000FF758E70E Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:11.974:
Skinny StationAlarmMessage on socket [2] 1.4.188.72 ATA000FF758E70E
*Apr 5 02:50:11.974: severityInformational p1=0 [0x0] p2=0 [0x0]
*Apr 5 02:50:11.974: Name=ATA000FF758E70E Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:12.066: ephone-(30)[2] StationRegisterMessage (29/31/48) from 1.4.188.72
*Apr 5 02:50:12.066: ephone-(30)[2] Register StationIdentifier DeviceName ATA000FF758E70E
*Apr 5 02:50:12.070: ephone-(30)[2] StationIdentifier Instance 1 deviceType 12
*Apr 5 02:50:12.070: ephone-30[-1]:stationIpAddr 1.4.188.72
*Apr 5 02:50:12.070: ephone-30[-1]:maxStreams 0
*Apr 5 02:50:12.070: ephone-30[-1]:protocol Ver 0x1
*Apr 5 02:50:12.070: ephone-30[-1]:phone-size 5392 dn-size 632
*Apr 5 02:50:12.070: ephone-(30) Allow any Skinny Server IP address 1.4.188.65
*Apr 5 02:50:12.070: ephone-30[-1]:Found entry 29 for 000FF758E70E
*Apr 5 02:50:12.070: ephone-30[-1]:socket change -1 to 2
*Apr 5 02:50:12.070: ephone-30[-1]:FAILED: CLOSED old socket -1
*Apr 5 02:50:12.074: ephone-30[2]:phone ATA000FF758E70E re-associate OK on socket [2]
21:21:49: %IPPHONE-6-REGISTER: ephone-30:ATA000FF758E70E IP:1.4.188.72 Socket:2 DeviceType:Phone has registered.
*Apr 5 02:50:12.074: Phone 29 socket 2
*Apr 5 02:50:12.074: Phone 29 socket 2: Running Bravo ??
*Apr 5 02:50:12.074: Skinny Local IP address = 1.4.188.65 on port 2000
*Apr 5 02:50:12.074: Skinny Phone IP address = 1.4.188.72 15325
*Apr 5 02:50:12.074: ephone-30[2]:Signal protocol ver 8 to phone with ver 1
*Apr 5 02:50:12.074: ephone-30[2]:Date Format M/D/Y
*Apr 5 02:50:12.078: ephone-30[2]:RegisterAck sent to ephone 2: keepalive period 30 use sccp-version 1
*Apr 5 02:50:12.078: ephone-30[2]:CapabilitiesReq sent
*Apr 5 02:50:12.090: ephone-30[2]:VersionReq received
*Apr 5 02:50:12.090: ephone-30[2]:Version String not needed for ATA device. Part of XML file
*Apr 5 02:50:12.090: ephone-30[2]:Version Message sent
*Apr 5 02:50:12.094: ephone-30[2]:CapabilitiesRes received
*Apr 5 02:50:12.098: ephone-30[2]:Caps list 7
G711Ulaw64k 60 ms
G711Alaw64k 60 ms
G729 60 ms
G729AnnexA 60 ms
G729AnnexB 60 ms
G729AnnexAwAnnexB 60 ms
Unrecognized Media Type 257 60 ms
*Apr 5 02:50:12.098: ephone-30[2]:ButtonTemplateReqMessage
*Apr 5 02:50:12.098: ephone-30[2]:StationButtonTemplateReqMessage set max presentation to 2
*Apr 5 02:50:12.098: ephone-30[2]:CheckAutoReg
*Apr 5 02:50:12.102: ephone-30[2]:AutoReg is disabled
*Apr 5 02:50:12.102: ephone-30[2][ATA000FF758E70E]:Setting 1 lines 4 speed-dials on phone (max_line 2)
*Apr 5 02:50:12.102: ephone-30[2]:First Speed Dial Button location is 2 (0)
*Apr 5 02:50:12.102: ephone-30[2]:Configured 4 speed dial buttons
*Apr 5 02:50:12.102: ephone-30[2]:ButtonTemplate lines=1 speed=4 buttons=5 offset=0
*Apr 5 02:50:12.102: ephone-30[2]:Skinny IP port 16384 set for socket [2]
*Apr 5 02:50:12.126: ephone-30[2]:StationSoftKeyTemplateReqMessage
*Apr 5 02:50:12.126: ephone-30[2]:StationSoftKeyTemplateResMessage
*Apr 5 02:50:12.206: ephone-30[2]:StationSoftKeySetReqMessage
*Apr 5 02:50:12.206: ephone-30[2]:StationSoftKeySetResMessage
*Apr 5 02:50:12.307: ephone-30[2]:StationLineStatReqMessage from ephone line 1
*Apr 5 02:50:12.307: ephone-30[2]:StationLineStatReqMessage ephone line 1 DN 80 = 8080 desc = 8080 label =
*Apr 5 02:50:12.307: ephone-30[2][ATA000FF758E70E]:StationLineStatResMessage sent to ephone (1 of 2)
*Apr 5 02:50:12.427: ephone-30[2]:StationSpeedDialStatReqMessage speed 9
*Apr 5 02:50:12.427: ephone-30[2]:No speed-dial set 9
*Apr 5 02:50:12.427: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.547: ephone-30[2]:StationSpeedDialStatReqMessage speed 8
*Apr 5 02:50:12.547: ephone-30[2]:No speed-dial set 8
*Apr 5 02:50:12.547: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.635: ephone-30[2]:StationSpeedDialStatReqMessage speed 7
*Apr 5 02:50:12.635: ephone-30[2]:No speed-dial set 7
*Apr 5 02:50:12.635: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.707: New Skinny socket accepted [1] (34 active)
*Apr 5 02:50:12.707: sin_family 2, sin_port 15400, in_addr 1.4.188.72
*Apr 5 02:50:12.711: skinny_add_socket 1 1.4.188.72 15400
*Apr 5 02:50:12.711: ephone-30[2]:StationSpeedDialStatReqMessage speed 6
*Apr 5 02:50:12.711: ephone-30[2]:No speed-dial set 6
*Apr 5 02:50:12.715: ephone-30[2]:StationSpeedDialStatMessage sent
21:21:50: %IPPHONE-6-REG_ALARM: Name=ATA0FF758E70E01 Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:12.715:
Skinny StationAlarmMessage on socket [3] 1.4.188.72 ATA000FF758E70E
*Apr 5 02:50:12.715: severityInformational p1=0 [0x0] p2=0 [0x0]
*Apr 5 02:50:12.715: Name=ATA0FF758E70E01 Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:12.811: ephone-30[2]:StationSpeedDialStatReqMessage speed 5
*Apr 5 02:50:12.811: ephone-30[2]:No speed-dial set 5
*Apr 5 02:50:12.811: ephone-30[2]:StationSpeedDialStatMessage sent
21:21:50: %IPPHONE-6-REGISTER: ephone-31:ATA0FF758E70E01 IP:1.4.188.72 Socket:3 DeviceType:Phone has registered.
*Apr 5 02:50:12.908: ephone-30[2]:StationSpeedDialStatReqMessage speed 4
*Apr 5 02:50:12.908: ephone-30[2]:No speed-dial set 4
*Apr 5 02:50:12.908: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.008: ephone-30[2]:StationSpeedDialStatReqMessage speed 3
*Apr 5 02:50:13.008: ephone-30[2]:No speed-dial set 3
*Apr 5 02:50:13.008: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.108: ephone-30[2]:StationSpeedDialStatReqMessage speed 2
*Apr 5 02:50:13.108: ephone-30[2]:No speed-dial set 2
*Apr 5 02:50:13.108: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.208: ephone-30[2]:StationSpeedDialStatReqMessage speed 1
*Apr 5 02:50:13.208: ephone-30[2]:No speed-dial set 1
*Apr 5 02:50:13.208: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:14.626: New Skinny socket accepted [1] (33 active)
*Apr 5 02:50:14.626: sin_family 2, sin_port 15593, in_addr 1.4.188.72
*Apr 5 02:50:14.630: skinny_add_socket 1 1.4.188.72 15593
*Apr 5 02:50:15.628: New Skinny socket accepted [1] (34 active)
*Apr 5 02:50:15.628: sin_family 2, sin_port 15693, in_addr 1.4.188.72
*Apr 5 02:50:15.628: skinny_add_socket 1 1.4.188.72 15693
*Apr 5 02:50:21.538: ephone-30[2]:SkinnyCompleteRegistration
Using Call Pickup and Group Call Pickup with Cisco ATA
Most of the procedures for using Cisco ATAs with Cisco Unified CME are the same as those for using Cisco ATAs with Cisco Unified Communications Manager, as described in the "How to Use Pre-Call and Mid-Call Services" chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0). However, the call pickup and group call pickup procedures are different when using Cisco ATAs with Cisco Unified CME, as described below:
Call Pickup
When using Cisco ATAs with Cisco Unified CME:
•To pickup the last parked call, press **3*.
•To pickup a call on a specific extension, press **3 and enter the extension number.
•To pickup a call from a park slot, press **3 and enter the park slot number.
Group Call Pickup
When using Cisco ATAs with Cisco Unified CME:
•To answer a phone within your call pickup group, press **4*.
•To answer a phone outside of your call pickup group, press **4 and the group ID number.
Note If there is only one pickup group, you do not need to enter the group ID after the **4 to pickup a call.
SCCP: Configuring Analog Phone Support
Configuring Cisco Unified CME to support calls and features on analog endpoints is basically the same as configuring any SCCP phone in Cisco Unified CME. This section describes only the steps that have special meaning for SCCP analog phone support.
Prerequisites
•Cisco CME 3.2.2 or a later version for analog FXS ports on the Cisco VG 224 Voice Gateway.
•Cisco Unified CME 4.0 or a later version for analog FXS ports on the Cisco 2800 Series or Cisco 3800 Series Integrated Services Routers.
Restrictions
•FXS ports on Cisco VG 248 Analog Phone Gateways are not supported by Cisco Unified CME.
•You must set the transfer-system command to full-blind or full-consult to enable call transfer on analog endpoints.
•You must set the timeouts ringing command to infinity (default) on the analog ports to prevent this timeout from expiring before the ringing no-answer timeout that is configured on Cisco Unified CME with the timeouts ringing command in telephony-service mode.
Note In Cisco IOS Release 12.4(11)T and later the default value of the timeouts ringing command is set to infinity for all SCCP-controlled analog ports. In releases earlier than Cisco IOS Release 12.4(11)T, the default is 180 seconds.
SUMMARY STEPS
1. Set up ephone-dns for up to 24 analog endpoints on the Cisco IOS gateway.
2. Set the maximum number of ephones.
3. Assign ephone-dns to ephones.
4. Set up feature parameters as desired.
5. Set up feature restrictions as desired.
DETAILED STEPS
Step 1 Set up ephone-dns for up to 24 endpoints on the Cisco IOS gateway.
Use the ephone-dn command:
ephone-dn 1 dual-line
number 1000
.
.
.
ephone-dn 24 dual-line
number 1024
Step 2 Set the maximum number of ephones.
Use the max ephones command to set a number equal to or greater than the total number of endpoints that you intend to register on the Cisco Unified CME router, including both IP and analog endpoints. For example, if you have 6 IP phones and 12 analog phones, set the max ephones command to 18 or greater.
Step 3 Assign ephone-dns to ephones.
Use the auto assign command to enable the automatic assignment of an available ephone-dn to each phone as the phone contacts the Cisco Unified CME router to register. Note that the order of ephone-dn assignment is not guaranteed. For example, if you have analog endpoints on ports 2/0 through 2/23 on the Cisco IOS gateway, port 2/0 does not necessarily become ephone 1. Use one of the following commands to enable automatic ephone-dn assignment.
•auto assign 1 to 24—You do not need to use the type keyword if you have only analog endpoints to be assigned or if you want all endpoints to be automatically assigned.
•auto assign 1 to 24 type anl—Use the type keyword if you have other phone types in the system and you want only the analog endpoints to be assigned to ephone-dns automatically.
An alternative to using the auto assign command is to manually assign ephone-dns to ephones (analog phones on FXS ports). This method is more complicated, but you might need to use it if you want to assign a specific extension number (ephone-dn) to a particular ephone. The reason that manual assignment is more complicated is because a unique device ID is required for each registering ephone and analog phones do not have unique MAC addresses like IP phones do. To create unique device IDs for analog phones, the auto assign process uses a particular algorithm. When you make manual ephone assignments, you have to use the same algorithm for each phone that receives a manual assignment. Note that once you have assigned ephone-dns to all the ephones that you want to assign manually, you can use the auto assign command to automatically assign the remaining ports.
The algorithm uses the single 12-digit SCCP local interface MAC address on the Cisco IOS gateway as the base to create unique 12-digit device IDs for all the FXS ports on the Cisco IOS gateway. The rightmost 9 digits of the SCCP local interface MAC address are shifted left three places and are used as the leftmost 9 digits for all 24 individual device IDs. The remaining 3 digits are the hexadecimal translation of the binary representation of the port's slot number (3 digits), subunit number (2 digits), and port number (7 digits). The following example shows the use of the algorithm to create a unique device ID for one port:
1. The MAC address for the Cisco VG 224 SCCP local interface is 000C.8638.5EA6.
2. The FXS port has a slot number of 2 (010), a subunit number of 0 (00), and a port number of 1 (0000001). The binary digits are strung together to become 0100 0000 0001, which is then translated to 401 in hexadecimal to create the final device ID for the port and ephone.
3. The resulting unique device ID for this port is C863.85EA.6401.
When setting up an ephone manually in ephone configuration mode for an analog port, assign it just one button because the port represents a single-line device. The button command can use the ":" (colon, for normal), "o" (overlay) and "c" (call-waiting overlay) modes.
Step 4 Set up feature parameters as desired.
•Call transfer—To use call transfer from analog endpoints, the transfer-system command must be configured for full-blind or full-consult in telephony-service configuration mode on the Cisco CME router. This is the recommended setting for Cisco CME 3.0 and later versions, but it is not the default.
•Call forwarding—Call forwarding destinations are specified for all, busy, and no-answer conditions for each ephone-dn using the call-forward all, call-forward busy, and call-forward noan commands in ephone-dn configuration mode.
•Call park—Call-park slots are created using the park-slot command in ephone-dn configuration mode. Phone users must be instructed how to transfer calls to the call-park slots and use directed pickup to retrieve the calls.
•Call pickup groups—Extensions are added to pickup groups using the pickup-group command in ephone-dn configuration mode. Phone users must be told which phones are in which groups.
•Caller ID—Caller names are defined using the name command in ephone-dn configuration mode. Caller numbers are defined using the number command in ephone-dn configuration mode.
•Speed dial—Numbers to be speed-dialed are stored with their associated speed-dial codes using the speed-dial command in ephone configuration mode.
•Speed dial to voice mail—The voice-mail number is defined using the voicemail command in telephony-service configuration mode.
Step 5 Set up feature restrictions as desired.
Features such as transfer, conference, park, pickup, group pickup (gpickup), and call forward all (cfwdall) can be restricted from individual ephones using the Cisco Unified CME soft-key template customization command, even though analog phones do not have soft keys. Simply create a template that leaves out the soft key that represents the feature you want to restrict and apply the template to the ephone for which you want the feature restricted. For more information about soft-key template customization, see "Customizing Soft Keys" on page 875.
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see "SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G" on page 943.
•After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267.
SCCP: Verifying Analog Phone Support
Use the following show commands to display information about analog endpoints.
•show ephone anl—Displays MAC address, registration status, ephone-dn, and speed-dial numbers for analog ephones.
•show telephony-service ephone-dn—Displays call forward, call waiting, pickup group, and more information about ephone-dns.
•show running-config—Displays running configuration nondefault values.
SCCP: Enabling a Remote Phone
To enable IP phones or instances of Cisco IP Communicator to connect to a Cisco Unified CME system over a WAN, perform the following steps.
Prerequisites
•The WAN link supporting remote teleworker phones should be configured with a Call Admission Control (CAC) or Resource Reservation Protocol (RSVP) solution to prevent the oversubscription of bandwidth, which can degrade the quality of all voice calls.
•If DSP farms will be used for transcoding, you must configure them separately. See "Configuring Transcoding Resources" on page 323.
•A SCCP phone to be enabled as a remote phone is configured in Cisco Unified CME. For configuration information, see the "SCCP: Creating Directory Numbers" section
Restrictions
•Because Cisco Unified CME is not designed for centralized call processing, remote phones are supported only for fixed teleworker applications, such as working from a home office.
•Cisco Unified CME does not support CAC for remote SCCP phones, so voice quality can degrade if a WAN link is oversubscribed. High-bandwidth data applications used over a WAN can cause degradation of voice quality for remote IP phones.
•Cisco Unified CME does not support Emergency 911 (E911) calls from remote IP phones. Teleworkers using remote phones connected to Cisco Unified CME over a WAN should be advised not to use these phones for E911 emergency services because the local public safety answering point (PSAP) will not be able to obtain valid calling-party information from them.
We recommend that you make all remote phone users aware of this issue. One way is to place a label on all remote teleworker phones that reminds users not to place 911 emergency calls on remote IP phones. Remote workers should place any emergency calls through locally configured hotel, office, or home phones (normal land-line phones) whenever possible. Inform remote workers that if they must use remote IP phones for emergency calls, they should be prepared to provide specific location information to the answering PSAP personnel, including street address, city, state, and country.
SUMMARY STEPS
1. enable
2. configure terminal
3. ephone phone-tag
4. mtp
5. codec {g711ulaw | g729r8 [dspfarm-assist]}
6. end
DETAILED STEPS
What to Do Next
•If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the "Configuring Codec for Local Calling Between SIP and SCCP Phones" section.
•To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see "SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G" on page 943.
•After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate configuration files for the phones to be connected. See "SCCP: Generating Configuration Files for SCCP Phones" on page 267.
SCCP: Verifying Remote Phones
Step 1 Use the show running-config command or the show telephony-service ephone command to verify parameter settings for remote ephones.
SCCP: Configuring Cisco IP Communicator Support
To enable support for Cisco IP Communicator, perform the following steps.
Prerequisites
•Cisco Unified CME 4.0 or a later version
•Cisco IP Communicator 2.0 or a later version
•IP address of the Cisco Unified CME TFTP server
•(Optional) Headsets with microphones for users
DETAILED STEPS
Step 1 Download the latest version of the Cisco IP Communicator software and install it on your PC.
The download website is at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
Step 2 (Optional) Attach a headset with microphone to your PC.
Step 3 Start the Cisco IP Communicator application.
Step 4 Define the IP address of the Cisco Unified CME TFTP server.
a. Open the Network > User Preferences window.
b. Enter the IP address of the Cisco Unified CME TFTP server.
Step 5 Wait for the Cisco IP Communicator application to connect to Cisco Unified CME and register.
Step 6 Configure the extension numbers and line buttons for the Cisco IP Communicator.
Use the normal phone provisioning commands described in the "SCCP: Creating Directory Numbers" section. In the type command, use the CIPC keyword to identify this phone as a Cisco IP Communicator.
SCCP: Verifying Cisco IP Communicator Support
Step 1 Use the show running-config command to display ephone-dn and ephone information associated with this phone.
Step 2 After Cisco IP Communicator registers with Cisco Unified CME, it displays the phone extensions and soft keys in its configuration. Verify that these are correct.
Step 3 Make a local call from the phone and have someone call you. Verify that you have a two-way voice path.
SCCP: Troubleshooting Cisco IP Communicator Support
Step 1 Use the debug ephone detail command to diagnose problems with calls. For more information, see the Cisco Unified CME Command Reference.
Configuration Examples for Making Basic Calls
This section contains the following examples of the required Cisco Unified CME configurations with some of the additional options that are discussed in other modules.
• Configuring SCCP Phones for Making Basic Calls: Example
• Configuring SIP Phones for Making Basic Calls: Example
• Disabling a Bulk Registration for a SIP Phone: Example
• Remote Teleworker Phones: Example
Configuring SCCP Phones for Making Basic Calls: Example
Router# show running-config
version 12.4
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME40
!
boot-start-marker
boot-end-marker
!
logging buffered 2000000 debugging
!
no aaa new-model
!
resource policy
!
clock timezone PST -8
clock summer-time PDT recurring
no network-clock-participate slot 2
voice-card 0
no dspfarm
dsp services dspfarm
!
voice-card 2
dspfarm
!
no ip source-route
ip cef
!
!
!
ip domain name cisco.com
ip multicast-routing
!
!
ftp-server enable
ftp-server topdir flash:
isdn switch-type primary-5ess
!
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
no supplementary-service h450.2
no supplementary-service h450.3
h323
call start slow
!
!
!
controller T1 2/0/0
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2/0/1
framing esf
linecode b8zs
!
!
interface GigabitEthernet0/0
ip address 192.168.1.1 255.255.255.0
ip pim dense-mode
duplex auto
speed auto
media-type rj45
negotiation auto
!
interface Service-Engine1/0
ip unnumbered GigabitEthernet0/0
service-module ip address 192.168.1.2 255.255.255.0
service-module ip default-gateway 192.168.1.1
!
interface Serial2/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn incoming-voice voice
isdn map address ^.* plan unknown type international
no cdp enable
!
!
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip route 192.168.1.2 255.255.255.255 Service-Engine1/0
ip route 192.168.2.253 255.255.255.255 10.2.0.1
ip route 192.168.3.254 255.255.255.255 10.2.0.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
tftp-server flash:P00307020300.loads
tftp-server flash:P00307020300.sb2
tftp-server flash:P00307020300.sbn
!
control-plane
!
!
!
voice-port 2/0/0:23
!
!
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.1.1 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register MTP0013c49a0cd0
keepalive retries 5
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
maximum sessions 90
associate application SCCP
!
!
dial-peer voice 9000 voip
mailbox-selection last-redirect-num
destination-pattern 78..
session protocol sipv2
session target ipv4:192.168.1.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 pots
incoming called-number .
direct-inward-dial
port 2/0/0:23
forward-digits all
!
dial-peer voice 1 pots
destination-pattern 9[2-9]......
port 2/0/0:23
forward-digits 8
!
dial-peer voice 3 pots
destination-pattern 91[2-9]..[2-9]......
port 2/0/0:23
forward-digits 12!
!
gateway
timer receive-rtp 1200
!
!
telephony-service
load 7960-7940 P00307020300
max-ephones 100
max-dn 300
ip source-address 192.168.1.1 port 2000
system message CCME 4.0
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 MTP0013c49a0cd0
voicemail 7800
max-conferences 24 gain -6
call-forward pattern .T
moh music-on-hold.au
multicast moh 239.1.1.1 port 2000
web admin system name admin password sjdfg
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn-template 1
!
!
ephone-template 1
keep-conference endcall local-only
codec g729r8 dspfarm-assist
!
!
ephone-template 2
!
!
ephone-dn 1
number 6001
call-forward busy 7800
call-forward noan 7800 timeout 10
!
!
ephone-dn 2
number 6002
call-forward busy 7800
call-forward noan 7800 timeout 10
!
!
ephone-dn 10
number 6013
paging ip 239.1.1.1 port 2000
!
!
ephone-dn 20
number 8000....
mwi on
!
!
ephone-dn 21
number 8001....
mwi off
!
!
!
!
ephone 1
device-security-mode none
username "user1"
mac-address 002D.264E.54FA
codec g729r8 dspfarm-assist
type 7970
button 1:1
!
!
!
ephone 2
device-security-mode none
username "user2"
mac-address 001C.821C.ED23
type 7960
button 1:2
!
!
!
line con 0
stopbits 1
line aux 0
stopbits 1
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output all
line 258
no activation-character
no exec
transport preferred none
transport input all
transport output all
line vty 0 4
exec-timeout 0 0
privilege level 15
password sgpxw
login
!
scheduler allocate 20000 1000
ntp server 192.168.224.18
!
!
end
Configuring SIP Phones for Making Basic Calls: Example
The following is a configuration example for SIP phones running on Cisco Unified CME:
voice service voip
allow-connections sip to sip
sip
registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
voice hunt-group 1 parallel
final 8000
list 2000,1000,2101
timeout 20
pilot 9000
voice hunt-group 2 sequential
final 1000
list 2000,2300
timeout 25
pilot 9100 secondary 9200
voice hunt-group 3 peer
final 2300
list 2100,2200,2101,2201
timeout 15
hops 3
pilot 9300
preference 5
voice hunt-group 4 longest-idle
final 2000
list 2300,2100,2201,2101,2200
timeout 15
hops 5
pilot 9400 secondary 9444
preference 5 secondary 9
voice register global
mode cme
external-ring bellcore-dr3
voice register dn 1
number 2300
mwi
voice register dn 2
number 2200
call-forward b2bua all 1000
call-forward b2bua mailbox 2200
mwi
voice register dn 3
number 2201
after-hour exempt
voice register dn 4
number 2100
call-forward b2bua busy 2000
mwi
voice register dn 5
number 2101
mwi
voice register dn 76
number 2525
call-forward b2bua unreachable 2300
mwi
!
voice register template 1
!
voice register template 2
no conference enable
voicemail 7788 timeout 5
!
voice register pool 1
id mac 000D.ED22.EDFE
type 7960
number 1 dn 1
template 1
preference 1
no call-waiting
codec g711alaw
!
voice register pool 2
id mac 000D.ED23.CBA0
type 7960
number 1 dn 2
number 2 dn 2
template 1
preference 1
dtmf-relay rtp-nte
speed-dial 3 2001
speed-dial 4 2201
!
voice register pool 3
id mac 0030.94C3.053E
type 7960
number 1 dn 3
number 3 dn 3
template 2
!
voice register pool 5
id mac 0012.019B.3FD8
type ATA
number 1 dn 5
preference 1
dtmf-relay rtp-nte
codec g711alaw
voice register pool 6
id mac 0012.019B.3E88
type ATA
number 1 dn 6
number 2 dn 7
template 2
dtmf-relay-rtp-nte
call-forward b2bua all 7778
voice register pool 7
voice register pool 8
id mac 0006.D737.CC42
type 7940
number 1 dn 8
template 2
preference 1
codec g711alaw
voice-port 1/0/0
voice-port 1/0/1
dial-peer voice 100 pots
destination-pattern 2000
port 1/0/0
dial-peer voice 101 pots
destination-pattern 2010
port 1/0/1
dial-peer voice 1001 voip
preference 1
destination-pattern 1...
session protocol sipv2
session target ipv4:10.15.6.13
codec g711ulaw
sip-ua
mwi-server ipv4:1.15.6.200 expires 3600 port 5060 transport udp
telephony-service
load 7960-7940 P0S3-07-2-00
max-ephones 24
max-dn 96
ip source-address 10.15.6.112 port 2000
create cnf-files version-stamp Aug 24 2004 00:00:00
max-conferences 8
after-hours block pattern 1 1...
after-hours day Mon 17:00 07:00
Disabling a Bulk Registration for a SIP Phone: Example
The following example shows the configuration for all phone numbers that match the pattern "408555.." can register with the SIP proxy server (IP address 1.5.49.240) except directory number 1, number "4085550101," for which bulk registration is disabled
voice register global
mode cme
bulk 408555....
voice register dn 1
number 4085550101
no-reg
sip-ua
registrar ipv4:1.5.49.240
Cisco ATA: Example
The following example shows the configuration for two analog phones using a single Cisco ATA with MAC address 000F.F758.E70E. The analog phone attached to the first port uses the MAC address of the Cisco ATA. The analog phone attached to the second port uses a modified version of the Cisco ATA's MAC address; the first two hexadecimal numbers are removed and 01 is appended to the end.
!
telephony-service
conference hardware
load ATA ATA030203SCCP051201A.zup
!
ephone-dn 80 dual-line
number 8080
!
ephone-dn 81 dual-line
number 8081
!
ephone 30
mac-address 000F.F758.E70E
type ata
button 1:80
!
ephone 31
mac-address 0FF7.58E7.0E01
type ata
button 1:81
SCCP Analog Phone: Example
The following excerpt from a Cisco Unified CME configuration sets transfer type to full-blind and sets the voice-mail extension to 5200. Ephone-dn 10 has the extension 4443 and is assigned to Tommy; that number and name will be used for caller-ID displays. The description field under ephone-dn is used to indicate that this ephone-dn is on the Cisco VG 224 voice gateway at port 1/3. Extension 4443 is assigned to ephone 7, which is an analog phone type with 10 speed-dial numbers.
CME_Router# show running-config
.
.
.
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
load 7905 CP79050101SCCP030530B31
max-ephones 60
max-dn 60
ip source-address 10.8.1.2 port 2000
auto assign 1 to 60
create cnf-files version-stamp 7960 Sep 28 2004 17:23:02
voicemail 5200
mwi relay
mwi expires 99999
max-conferences 8 gain -6
web admin system name cisco password lab
web admin customer name ac2 password cisco
dn-webedit
time-webedit
transfer-system full-blind
transfer-pattern 6...
transfer-pattern 5...
!
!
ephone-dn 10 dual-line
number 4443 secondary 9191114443
pickup-group 5
description vg224-1/3
name tommy
!
ephone 7
mac-address C863.9018.0402
speed-dial 1 4445
speed-dial 2 4445
speed-dial 3 4442
speed-dial 4 4441
speed-dial 5 6666
speed-dial 6 1111
speed-dial 7 1112
speed-dial 8 9191114441
speed-dial 9 9191114442
speed-dial 10 9191114442
type anl
button 1:10
!
Remote Teleworker Phones: Example
The following example shows the configuration for ephone 270, a remote teleworker phone with its codec set to G.729r8. The dspfarm-assist keyword is used to ensure that calls from this phone will use DSP resources to maintain the G.729r8 codec when calls would normally be switched to a G.711 codec.
ephone 270
button 1:36
mtp
codec g729r8 dspfarm-assist
description teleworker remote phone
Where to Go Next
To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see "SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G" on page 943 .
After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate configuration files for the phones to be connected to your router. See "Generating Configuration Files for Phones" on page 265.
Additional References
The following sections provide references related to Cisco Unified CME features.
Related Documents
Related Topic Document TitleCisco Unified CME configuration
• Cisco Unified CME Command Reference
Cisco IOS commands
• Cisco IOS Voice Command Reference
Cisco IOS configuration
• Cisco IOS Voice Configuration Library
Phone documentation for Cisco Unified CME
Technical Assistance
Feature Information for Configuring Phones to Make Basic Calls
Table 11 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Note Table 11 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.
Posted: Tue Sep 25 13:42:16 PDT 2007
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