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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
Table 11-5 North America PSTN Dial Plan
Route Pattern Description
911 Emergency call routing without access code
9.911 Emergency call routing with access code
9.[2–8]11 3-digit service codes (for example, 411 for information)
9.[2–9]XX XXXX 7-digit local dialing
9.[2–9]XX [2–9] XXXX 10-digit local dialing
9.1[2–9]XX [2–9]XX XXXX 11-digit long-distance dialing
9.011! International dialing (variable-length dialing)
9.011!# International dialing (variable-length dialing with interdigit
termination)
Matches 100-Digit Strings
Matches Digit Strings
Matches 10-Digit Strings
Select as Closest Match
1111
121X
1[23]XX
131
13[0-4]X
13!
Does Not Match
Match!
Match!
Match!
Does Not Match
User Dial String:
1311<timeout>
Does Not Match
Digit Forwarding 265
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.
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.
Table 11-6 Digit-Forwarding Behavior
Device Signaling Protocol Addressing Method
IP phone SCCP Digit by digit
SIP Enbloc
KPML
SIP dial rules
Gateway MGCP/SIP/H.323 Enbloc
Overlap sending and receiving (ISDN PRI only)
Trunk H.323/SIP Enbloc
Overlap sending and receiving (ISDN PRI only)
266 Chapter 11: Call Routing Components
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
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.
IP
Off-Hook, Digit 1, Digit 0, Digit 0, Digit 0
Dial Tone On/Off, Screen Update, etc.
Dialing Actions:
1000
SCCP Message Sent
with Each User Action
Dial Plan
(Digit Analysis)
Any Phone
Model
Running
SCCP
Signaling
Digit Forwarding 267
Figure 11-8 SIP Type A Phones: No Dial Rules
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.
IP
“Call for 1000”
Call in Progress, Call Connected, Call Denied, etc.
Dialing Actions:
1000 Dial
SIP INVITE Message Sent
when User Presses the Dial Key
Dial Plan
(Digit Analysis)
Existing
SIP Phone
Such as
7940, 7960
Signaling
268 Chapter 11: Call Routing Components
Figure 11-9 SIP Type A Phones: Dial Rules
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
IP
Off-Hook, Digit 1, Digit 0, Digit 0, Digit 0
Call in Progress, Call Connected, Call Denied, etc.
Dialing Actions:
1000
KPML Events Reported
in SIP NOTIFY Messages
Dial Plan
(Digit Analysis)
SIPEnhanced
Phone
Such as
7971
Signaling
IP
“Call for 1000”
Call in Progress, Call Connected, Call Denied, etc.
Dialing Actions:
1000
Pattern 1...
Timeout 0
SIP INVITE Message
Sent When Pattern Is Recognized
Dial Plan
(Digit Analysis)
SIPEnhanced
Phone
Such as
7971
Signaling
CUCM Path Selection 269
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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