Digit Forwarding

Table 11-6 displays a summary of the digit-forwarding methods supported in CUCM for different types of devices.

SIP devices support enbloc dialing by default. Enbloc dialing sends the entire dialed string in a single SIP INVITE message. KPML is an IETF SIP standards-based extension that Cisco supports. KPML allows digits to be sent one by one. Although Cisco supports KPML, multivendor interoperability might prove difficult because most vendors do not support this standard. SIP dial rules offer yet another option that can be used in SIP devices. SIP dial rules are downloaded to the phone and processed inside the SIP phone. A SIP phone can detect invalid numbers and play a reorder tone without sending any signaling messages to CUCM.

Trunks and ISDN PRIs send their digits enbloc by default, but they can both be configured for overlap sending and receiving, allowing digits to be sent or received one by one over an ISDN PRI.

Table 11-6 Digit-Forwarding Behavior

Device

Signaling Protocol

Addressing Method

IP phone

SCCP

Digit by digit

SIP

Enbloc KPML SIP dial rules

Gateway

Overlap sending and receiving (ISDN PRI only)

Trunk

Overlap sending and receiving (ISDN PRI only)

SCCP Phones: User Input

IP phones using SCCP report every user input event to CUCM immediately. As soon as the user goes off-hook, a signaling message is sent from the phone to the CUCM server with which it is registered. The phone can be considered to be a terminal, where all decisions resulting from the user input are made by dial plan configured on the CUCM server.

As other user events are detected by the phone, they are relayed to CUCM individually. A user who goes off-hook and then dials 1000 triggers five individual signaling events from the phone to CUCM. All the resulting feedback provided to the user (screen messages, playing dial tone, secondary dial tone, ringback, reorder, and so forth) are commands issued by CUCM to the phone in response to the dial plan configuration.

It is neither required nor possible to configure dial plan information on IP phones running SCCP. All dial plan functionality is contained in the CUCM cluster, including the recognition of dialing patterns as user input is collected.

If the user dials a pattern that is denied by CUCM, a reorder tone is played to the user as soon as that pattern becomes the best match in CUCM digit analysis. For example, if all calls to the 976 area code are denied, a reorder tone is sent to the user's phone as soon as the user dials 91976.

Figure 11-7 SCCP Phones: User Input

Any Phone Model Running SCCP

Any Phone Model Running SCCP

Dialing Actions: 1000

SCCP Message Sent with Each User Action

Dial Plan (Digit Analysis)

Off-Hook, Digit 1, Digit 0, Digit 0, Digit 0 Dial Tone On/Off, Screen Update, etc.

Dialing Actions: 1000

Signaling

Dial Plan (Digit Analysis)

SIP Phones: User Input

Type A phones (Cisco Unified IP Phone models 7905, 7912, 7940, and 7960) do not support KPML. They do support SIP dial rules, which are configured in CUCM and downloaded to the IP phone at boot time.

Type B phones (Cisco Unified IP Phone models 7911, 7941, 7961, 7970, and 7971) support KPML and SIP dial rules.

Type A SIP Phones: No Dial Rules

Type A phones without SIP dial rules (default) do not deliver a dial tone to the calling party when the calling party goes off-hook with the handset, speakerphone, or headset. All digits are sent after the user completes dialing and clicks the Dial softkey. This function is similar to the Send button used on cellular phones.

Figure 11-8 illustrates a user making a call to extension 1000. The user has to dial 1000 followed by clicking the Dial softkey or the # key. The phone then sends a SIP INVITE message to CUCM for digit analysis.

Figure 11-8 SIP Type A Phones: No Dial Rules

SIP INVITE Message Sent when User Presses the Dial Key

Dial Plan (Digit Analysis)

Existing SIP Phone Such as 7940, 7960

Existing SIP Phone Such as 7940, 7960

Dialing Actions: 1000 Dial

Signaling

Dialing Actions: 1000 Dial

Signaling

Type A SIP Phones: Dial Rules

SIP dial rules allow SIP phones to emulate the functionality of a SCCP phone. When the user goes off-hook, a dial tone is received, and digits are processed against the local SIP dial rule in real time. If a user dials a pay service beginning with 9 1-900, the call is immediately dropped. Users are accustomed to hearing a reorder tone when a call cannot be routed. SIP dial rule pattern rejection does not result in a reorder tone. Whereas the loss of reorder tone might be seen as a deficit, SIP dial rules have positive network bandwidth and CUCM processor overhead advantages. SCCP uses many small signaling messages sent between the IP phone and CUCM. These constant SCCP messages result in delay when the IP phone and CUCM are separated by large geographical boundaries. The SCCP messages also use up a small amount of bandwidth across the expensive WAN data circuits and utilize processor and memory overhead on CUCM. SIP dial rules eliminate the need to send signaling across the network between the IP phone and CUCM.

When a permitted call occurs on the phone, the SIP INVITE message is sent enbloc to CUCM. The user does not need to press the Dial key. If you do not use SIP dial rules or KPML, the end-user community will have to be retrained, because the phone will need to be operated differently. SIP dial rules allow Type A phones to emulate SCCP and traditional phone systems, while also providing processing and signaling benefits.

Figure 11-9 shows a phone configured to recognize all four-digit patterns beginning with 1 and that has an associated timeout value of 0. All user input actions matching the pattern will trigger the sending of the SIP INVITE message to CUCM immediately, without requiring the user to press the Dial key. Type A phones using SIP dial rules offer a way to dial patterns not explicitly configured on the phone. If a dialed pattern does not match a SIP dial rule, the user can press the Dial key or wait for interdigit timeout.

If a particular pattern is recognized by the phone but blocked in the dial rule, the call is immediately ended. The user will not receive a reorder tone, but the session will end.

Figure 11-9 SIP Type A Phones: Dial Rules

KPML Events Reported in SIP NOTIFY Messages

Dial Plan (Digit Analysis)

SIP-Enhanced Phone Such as 7971

SIP-Enhanced Phone Such as 7971

Dialing Actions: 1000

Call in Progress, Call Connected, Call Denied, etc.

Off-Hook, Digit 1, Digit 0, Digit 0, Digit 0

Call in Progress, Call Connected, Call Denied, etc.

Signaling

Type B SIP Phones: No Dial Rules

Type B IP phones offer functionality based on the KPML standard to report user activities. Each user input event generates a KPML message to CUCM. This mode of operation emulates a similar end-user experience to that of phones running SCCP.

Every key the end user presses triggers an individual SIP message. SIP NOTIFY messages are sent to CUCM to report a KPML event corresponding to the key pressed by the user. This messaging enables CUCM digit-by-digit analysis to recognize partial patterns as the user dials them. If a pattern beginning with 9 1-900 is blocked, a reorder tone is sent to the calling party.

Users of Type B SIP phones do not need to click the Dial softkey to indicate the end of user input. In Figure 11-10, a user dialing 1000 would be provided call progress indication (either a ringback tone or reorder tone) after dialing the last 0, without having to press the Dial softkey. This behavior is consistent with the user experience of phones running SCCP.

Figure 11-10 SIP Type B Phones: No SIP Dial Rule

SIP INVITE Message Sent When Pattern Is Recognized

Dial Plan (Digit Analysis)

SIP-Enhanced Phone Such as 7971

SIP-Enhanced Phone Such as 7971

Dialing Actions: 1000

Pattern 1... Timeout 0

Signaling

Dialing Actions: 1000

Pattern 1... Timeout 0

Signaling

Type B SIP Phones: Dial Rules

Type B IP phones can be configured with SIP dial rules so that dial pattern recognition is accomplished by the phone.

Type B IP phones using SIP dial rules offer only one way to dial patterns not explicitly configured on the phone. If a dialed pattern does not match a SIP dial rule, the user has to wait for the interdigit timeout before the SIP NOTIFY message is sent to CUCM. Unlike Type A IP phones, Type B IP phones do not need the Dial softkey clicked to indicate the end of dialing. When on-hook dialing is used, the user can click the Dial softkey at any time to trigger the sending of all dialed digits to CUCM in one SIP INVITE message.

If a particular pattern is permitted by the phone but blocked by CUCM, the user must dial the entire dial string before receiving an indication that the call is rejected by the system. Dial rules should be configured to be more restrictive than the calling restrictions applied at the CUCM call-processing layer.

In countries whose national numbering plan is not easily defined with static route patterns, CUCM can be configured for overlap sending and overlap receiving. Overlap sending changes the way gateways pass digits on Q.931 gateways. To enable overlap sending, check the Allow Overlap Sending box on the Route Pattern Configuration page.

Overlap receiving allows CUCM to receive dialed digits one by one from a PRI PSTN gateway. To enable overlap receiving, set the OverlapReceivingFlagForPRI service parameter to True.

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