Cisco Systems Frozen Dessert Maker OL 12397 13 User Manual

C H A P T E R  
2
SIP Subscribers  
Revised: April 17, 2008, OL-12397-13  
The Cisco BTS 10200 Softswitch supports SIP subscribers on SIP phones that are compliant with  
RFC 3261 or RFC 2543. This section describes the support for SIP subscribers and how to provision SIP  
subscriber features.  
In this document  
SIP subscriber means a SIP phone that is registered directly to the BTS 10200 and for which the  
BTS 10200 maintains subscriber information.  
SIP ANI-based subscriber means a SIP phone that communicates with the BTS 10200 over a SIP  
trunk.  
Note  
For quick-reference tables listing the subscriber features, see the “Comparison of SIP-Based Features  
This section covers the following topics:  
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Chapter 2 SIP Subscribers  
SIP Registration and Security  
When a SIP user attempts to register or set up a call, the BTS 10200 challenges the SIP subscriber based  
on provisioning in the Serving Domain Name table. If the Serving Domain Name table indicates that  
authentication is required, the BTS 10200 challenges the SIP request (Register/INVITE) according to  
the authentication procedures specified in the SIP Protocol RFC 3261. If the BTS 10200 receives valid  
credentials, the authenticated AOR from the User Authorization table identifies the subscriber based on  
the Address of Record to Subscriber table. (For specific provisioning parameters, see the applicable  
Registration creates bindings in the BTS 10200 that associate an AOR with one or more contact  
addresses.  
The registration data is replicated on the standby BTS 10200. The BTS 10200 imposes a minimum  
registration interval as a provisionable value. If the expiration duration of the incoming registration  
request is lower than the provisioned minimum, a 423 (Interval Too Brief) response is sent to the  
registering SIP endpoint.  
The BTS 10200 generates a warning event when a request from a client fails authentication. This can  
indicate a provisioning error or an attempt by an unauthorized client to communicate with the  
BTS 10200.  
The contacts registered for an AOR can be looked up using the status command, as demonstrated by the  
following example.  
CLI>status sip-reg-contact [email protected]  
AOR ID -> [email protected]  
USER -> 4695551884  
HOST -> 10.88.11.237  
PORT -> 5060  
USER TYPE -> USER_PHONE_TYPE  
EXPIRES -> 3600  
EXPIRETIME -> Thu Jan 22 14:33:36 2004  
STATUS -> REGISTERED CONTACT  
Reply :Success:  
Enhanced SIP Registration  
SIP Registration ensures that a SIP REGISTER message to the BTS 10200 is from a provisioned  
endpoint, that is, an endpoint with a provisioned secure Fully-Qualified Domain Name (FQDN) or IP  
address. The feature also ensures that the source IP address and contact parameter for all originating calls  
are from the provisioned SIP endpoint, and that no calls can originate from an unregistered endpoint.  
Description  
Prior to Release 4.5.1, SIP endpoint registration was based on AOR, UserID, and password; there was  
no verification of the origination of the REGISTER message. Certain service providers may prefer that  
the source IP address of SIP requests be verified against a provisioned FQDN of the endpoint to address  
the possibility of theft of VoIP service.  
The BTS 10200 can indicate SECURE_FQDN provisioning for specified SIP term-type subscribers.  
This indication consists of specifying an FQDN with the subscriber AOR. The FQDN is the  
address/location of the SIP endpoint and is added to the AOR table. The FQDN does not have a service  
port.  
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Chapter 2 SIP Subscribers  
SIP Registration and Security  
To enable or disable SECURE_FQDN on a successful registered subscriber  
1. Take AOR out of service to remove all registered contact.  
2. Enable or disable SECURE_FQDN for the subscriber.  
3. Bring AOR back In-Service.  
4. Reboot the ATA.  
A subscriber with the secure FQDN feature enabled has the following characteristics:  
One and only one AOR is associated with the endpoint.  
Does not have any static-contact associated with it.  
UserId and Password Authentication are supported.  
One FQDN (specified without service port).  
The DNS lookup of the FQDN should result in one and only one IP address.  
Cannot place or receive a call unless successfully registered.  
Example  
This example presents a case in which a VoIP subscriber (Subscriber 1) uses the following options for  
the user ID, password, and phone number:  
user-id-1  
password-1  
phone-no-1  
Without security, another VoIP subscriber, Subscriber 2, could access Subscriber 1’s information  
(perhaps by getting a Cisco ATA configuration file with the encryption key in clear text, and then getting  
the full configuration file with all the data). Subscriber 2 could then register to the BTS 10200 with  
Subscriber 1’s combination of user-id-1, password-1, and phone-no-1, as well as Subscriber 2’s own IP  
address. Without the secure FQDN feature, the Cisco BTS 10200 would accept this information unless  
specific measures were taken, and Subscriber 2 could steal service and make calls on behalf of  
Subscriber 1.  
Provisioning Commands  
This section shows the CLI commands you need to provision a secure fully qualified domain name  
(FQDN) of a SIP endpoint.  
Note  
Use this procedure to provision subscribers on the BTS 10200. The procedure does not cover the security  
of configuration files provisioned on the SIP adapter (for example, an ATA), which are the responsibility  
of the service provider.  
The SECURE_FQDN token is present in both the SUBSCRIBER and AOR2SUB tables. A non-null  
value in the field indicates that the SECURE_FQDN validations apply to all SIP messages received from  
the endpoint associated with that AOR.  
The SECURE_FQDN value can be specified on a subscriber only if the AOR for the subscriber is  
OOS. When an AOR is taken administratively OOS, its registered contacts are deleted.  
A static contact cannot be specified for a SECURE_FQDN subscriber. Any existing static contact  
record for an AOR must be deleted before the subscriber can be made a SECURE_FQDN SIP  
endpoint.  
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Chapter 2 SIP Subscribers  
SIP Registration and Security  
The SECURE_FQDN in the AOR2SUB table is stored both in the ORACLE database and the shared  
memory.  
AOR2SUB records cannot be added or deleted directly. To add AOR2SUB records, you must specify  
specify the AOR ID on a subscriber record.  
Provision a New SIP Subscriber  
To provision a new SIP subscriber with the secure FQDN feature, enter the following command.  
Step 1  
Note  
This command automatically adds a corresponding entry in the AOR2SUB table.  
add subscriber id=sub1; sub-profile-id=subpf1; category=individual;  
dn1=241-555-1018; term-type=SIP; aor-id=<aor-id of SIP adapter port for sub1>;  
secure-fqdn=<secure-fqdn of the SIP adapter>;  
Step 2  
(Optional) To provision an additional subscriber on the same SIP adapter, enter the following command:  
add subscriber id=sub2; sub-profile-id=subpf1; category=individual;  
dn1=241-555-1022; term-type=SIP; aor-id=<aor-id of SIP adapter port for sub2>;  
secure-fqdn=<secure-fqdn of the SIP adapter>;  
Note  
If there are multiple subscribers on a single SIP adapter (such as an ATA), these subscribers  
might share the same IP address. Therefore, you can provision all of these subscriber records  
with a single secure-fqdn, and in the DNS, this FQDN can point to the applicable IP address.  
The id, dn1, and aor-id tokens must have unique values for each subscriber.  
Enable or Disable Secure FQDN for an Existing Subscriber  
To enable or disable the secure FQDN feature for a successfully registered subscriber, enter the  
following commands:  
Step 1  
Step 2  
Take the AOR out of service (OOS). This command removes all registered contact.  
change aor2sub [email protected]; status=oos;  
To enable the secure FQDN feature for an existing subscriber, enter the following command:  
change subscriber id=sub1; secure-fqdn=ata-SYS41CA146.ipclab.cisco.com  
To disable the secure FQDN feature for an existing subscriber, enter  
change subscriber id=sub1; secure-fqdn=null  
Note  
If secure-fqdn is not provisioned for the subscriber, the system does not provide the secure  
FQDN feature to that subscriber. If secure-fqdn has previously been provisioned for the  
subscriber, setting secure-fqdn to null disables the feature.  
Step 3  
To bring the AOR back in service (INS), enter the following command:  
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SIP Registration and Security  
change aor2sub [email protected]; status=ins;  
Step 4  
Reboot the adapter device (such as ATA) for this subscriber.  
Operations  
The system performs the following checks. If any of the following conditions are not met, the request is  
rejected, and an alarm is generated.  
No Calls to or from an Unregistered Secure-Provision SIP Endpoint  
An unregistered secure-provision SIP endpoint cannot originate or receive calls.  
Third-Party Registrations for Secure FQDN Endpoint Not Allowed  
Third-party registrations for secure FQDN endpoints are not allowed.  
Cisco BTS 10200 Challenges Registration  
On receiving a REGISTER message from a secure-provision SIP endpoint, the BTS 10200 challenges  
the registration, asking for authentication. Verification of the resend REGISTER message with UserId  
and Password is as follows, after the UserId and Password are authenticated:  
Ensure that there is only one contact in the contact header.  
Ensure that the source IP address of the REGISTER message is the same IP address of the  
provisioned FQDN for that endpoint.  
Ensure that the IP address or the FQDN of the contact is the same as the provisioned FQDN for that  
endpoint.  
If any of these conditions are not met, registration is rejected and a security event and alarm is generated,  
indicating that the source of the registration is illegal.  
The contact address can verify all subsequent SIP request source IP address of the request from the  
endpoint until the registration expired or is deregistered.  
Registration Expires  
If the registration expires or the end point de-registers, the registration process in the “Cisco BTS 10200  
Challenges Registration” section on page 2-6 occurs before any new calls are accepted.  
Call Originates From or Terminates to a Secure-Provision SIP Endpoint  
When a call originates from or terminates to a secure-provision SIP endpoint  
1. The system authenticates the user ID and password on all messages requiring authentication.  
2. If the Contact header is available, the system ensures that only one contact is present, and that it has  
the same IP address or FDQN of the provisioned endpoint.  
3. All messages sent by the endpoint and the source IP address of the message must be the same as the  
internal cache contact address (for example, the cache contact address is the contact obtained during  
registration).  
4. Response from an endpoint that has a contact header must conform to the second item in this list.  
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Chapter 2 SIP Subscribers  
SIP Registration and Security  
Call Processing  
The SIP application in the BTS 10200 implements the secure provisioning feature for all incoming SIP  
messages (requests and responses) from SIP endpoints.  
When a SIP request message is received from a SIP endpoint and Auth_Rqed=Y for the serving domain,  
the request is challenged. When the request is resubmitted with credentials, the AOR of the authenticated  
SIP endpoint is used to perform the SECURE_FQDN validation, provided a SECURE_FQDN value is  
provisioned in the AOR2SUB record. If Auth_Reqd=N, the SECURE_FQDN validation is performed  
without the request being challenged.  
Validation  
The validation processing for a SIP request, that comes from a SIP endpoint provisioned with this  
feature, is as follows:  
1. The SECURE_FQDN validation occurs on every request (including CANCEL/ACK).  
2. The SECURE_FQDN is verified to have a DNS resolution, if it is a domain name. If there is no DNS  
resolution, a 500 Internal Server Error response is returned.  
3. The DNS resolution for the SECURE_FQDN is verified to yield a single IP address Secure-IP1.  
If the address is incorrect, a 500 Internal Server Error response is returned.  
4. The Source IP address of the packet is verified as identical to Secure-IP1.  
If the address is not identical, a 403 Forbidden response is returned.  
5. If the Request is a Register, it is verified to have a single Contact header.  
If there is not a single contact header, a 403 Forbidden response is returned.  
6. If the SIP request is an initial INVITE (including an INVITE resubmitted with credentials), it is  
verified that there is an unexpired registered contact for the AOR.  
If here is not an unexpired registered contact, a 403 Forbidden response is returned.  
7. When a Contact header is present, the Contact FQDN/IP address of the request is verified to yield a  
single IP address Secure-IP1.  
If it does not yield the proper address, a 500 Internal Server Error response is returned.  
8. The IP address of the Contact host is verified as identical to the IP address Secure-IP1 of the  
SECURE_FQDN.  
If the addresses are not identical, a 403 Forbidden response is returned.  
9. The provisioning of a static contact on a AOR is not disabled, but any provisioned value is ignored  
because of the SECURE_FQDN validation rules. A static contact is irrelevant for SECURE_FQDN  
AORs, since the SIP request is denied if no registered contact exists.  
10. The To and From header URLs in a REGISTER are verified to be identical, for SECURE_FQDN  
subscribers. This is to block third-party registration.  
Received SIP Response Message  
When a SIP response message is received from a SIP endpoint, the following occurs:  
1. The Source IP address of the packet is verified to be identical with the IP address of the Secure-IP1.  
If the addresses are not identical, the response is dropped. This has the same result as the non-receipt  
of that response, such as would happen with a call failure.  
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SIP Registration and Security  
2. When a Contact header is present on a reliable 1xx or 2xx response, the Contact FQDN/IP address  
of the response is verified to resolve to the Secure-IP1.  
If the address does not resolve properly, the response is dropped. This has the same result as the  
non-receipt of that response, such as would happen with a call failure.  
3. The response for a BYE sent by Cisco BTS 10200 is not validated. This is the least likely point in a  
call for theft.  
Rules for Sending a SIP INVITE Message from the BTS 10200  
When a SIP INVITE message is sent to a SIP endpoint, the following occurs:  
1. The INVITE is sent to the registered contact of the endpoint. If there is no registered contact or if  
the registered contact has expired, the INVITE is not sent and the call is declined.  
2. Any static contact provisioned for the subscriber is ignored.  
Note  
Provisioning of static contact is not allowed for secure SIP endpoints; therefore, this is merely due  
diligence.  
Validation of ACK Request  
When a SIP ACK message is received from a SIP endpoint, the following occurs:  
1. The ACK for a 200-class response is validated like any other SIP request.  
2. The ACK for a failure response (3xx or higher) is not validated.  
Measurements  
The following TMM counters are supported for secure FQDN violations:  
A SIA-SECURE_FQDN-VIOLATION-REQ counter is incremented when a SIP request fails the  
validation for secure SIP endpoints.  
A SIA-SECURE_FQDN-VIOLATION-RESP counter is incremented when a SIP response fails the  
validation for secure SIP endpoints.  
Note  
For a full list of measurements, see the Cisco BTS 10200 Softswitch Operations and Maintenance Guide.  
Events and Alarms  
A Warning event is raised when a SIP request or response fails the validation for secure SIP endpoints.  
The alarm has the following attributes:  
Type: SECURITY(6)  
DESCRIPTION: Secure SIP Endpoint Validation Failure  
SEVERITY: WARNING  
Note  
For a full list of events and alarms, see the Cisco BTS 10200 Softswitch Troubleshooting Guide.  
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SIP User Authentication  
SIP User Authentication  
The BTS 10200 can act as an authentication server. Authentication is enabled on the serving domain  
through provisioning.  
Whenever a SIP request is received from a SIP subscriber, the request is authenticated to ensure it is  
indeed from an identified user. Authentication also enables request authorization, because users may be  
authorized to perform only specific requests.  
The following examples are the functional scenarios in which authentication is required:  
1. When a SIP user registers a contact with the BTS 10200 Registrar using a REGISTER request.  
2. When a SIP user initiates a call using an INVITE request.  
3. When a SIP user sends any request in an ongoing call. Examples include  
Re-negotiation of the call parameters using a re-INVITE  
Terminating the call using a BYE  
Initiating a call transfer using a REFER  
4. When a SIP user sends a request outside a dialog. Example: OPTIONS.  
The following tables affect authentication for SIP subscribers:  
AOR  
Serving Domain  
Auth-Realm  
User-Auth  
See the Cisco BTS 10200 Softswitch CLI Database for more information about the tables.  
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Chapter 2 SIP Subscribers  
SIP Subscriber Calls  
Figure 2-2 shows how an incoming request is processed, and indicates the role of the Authentication  
Service in the BTS 10200.  
Figure 2-2  
Authentication and Processing of an Incoming Request (for Example, INVITE)  
Cisco BTS 10200  
Softswitch  
SIP Phone 2  
SIP Phone 1  
IP  
IP  
Invite  
401  
ACK  
Invite  
Invite  
200  
ACK  
200  
ACK  
BYE  
401  
BYE  
200  
BYE  
200  
The BTS 10200 validates the hostname of the ReqUri of every incoming SIP request against the list of  
names provisioned in the Serving-Domain-Name table. The BTS 10200 hostname used by devices (in  
the ReqUri), when they send requests to the BTS 10200, should be provisioned in the  
Serving-Domain-Name table of that BTS 10200. If a name is not provisioned (and therefore not found)  
in the Serving-Domain-Name table, the BTS 10200 rejects the SIP request with a “404 Not Found  
ReqUri Serving Domain” response.  
The BTS 10200 authenticates IP phones by using the MD5 digest defined in RFCs 3261 and 2617. The  
BTS 10200 verifies a user’s credentials on each SIP request from the user. For more information, see the  
User Authorization table in the Cisco BTS 10200 Softswitch CLI Database.  
SIP Subscriber Calls  
SIP subscribers must present valid credentials on a SIP INVITE message in order to place calls.  
The system allows SIP subscribers to call other SIP subscribers or SIP trunks connected to the  
BTS 10200. The provisioned dial plan determines whom a subscriber can call. A SIP subscriber can  
receive a call as long as the subscription’s registration is current, or a static registration has been  
provisioned.  
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Chapter 2 SIP Subscribers  
Provisioning Session Timers for SIP Subscribers  
Provisioning Session Timers for SIP Subscribers  
The system uses session timers to periodically refresh SIP sessions during call processing or in-progress  
calls. You can enable or disable session timers for calls to and from all SIP subscribers on the BTS 10200  
through the SUB_SESSION_TIMER_ALLOWED parameter in the ca-config table. They are disabled  
by default.  
Use the commands in this section to provision session timers for SIP subscribers. Session timer defaults  
for subscribers are defined by internal defaults. They can be adjusted through the commands shown in  
this section.  
Note  
For a detailed description of session timers, see “SIP Session Timers” section on page 4-7  
Adjust the session timer values in the sip-timer-profile table.  
Step 1  
Note  
The session duration field value is in seconds with a range of 100 to 7200.  
The minimum session duration field value is in seconds with a range of 100 to 1800.  
We recommend a value of at least 1800 for each of these fields.  
add sip_timer_profile id=<timer_profile_id>; session_expires_delta_secs=7200;  
min-se=1800;  
Step 2  
Step 3  
Enable session timers for SIP subscribers:  
add ca-config type=SUB_SESSION_TIMER_ALLOWED; datatype=BOOLEAN; value=Y;  
If not already done, add a default sip-timer-profile-id to the ca-config table:  
add ca_config type=SIP_TIMER_PROFILE_ID; datatype=STRING;  
value=<sip_timer_profile_id>;  
SIP Timer Values for SIP Subscribers  
Note  
This section describes how to provision SIP timer values for SIP subscribers. For a comprehensive listing  
of SIP timers, see Chapter 4, “SIP System Features.”  
You can customize SIP timers through the sip-timer-profile table. A record in this table can then be  
configured to apply to all subscribers switch-wide. The system operates with default SIP protocol timer  
values, as noted in the SIP specification. These default values are adequate for many installations. If  
customization is required, a sip-timer-profile table can be provisioned and associated with all calls.  
Use the following steps to provision the SIP timer values.  
Step 1  
Adjust the SIP timer values in the sip-timer-profile table if necessary (example shown).  
add sip-timer-profile id=<timer_profile_id>; timer-t1-milli=500;  
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Chapter 2 SIP Subscribers  
Diversion Indication for SIP Subscribers  
Step 2  
If not already done, add a default sip-timer-profile-id to the ca-config table:  
add ca-config type=sip_timer_profile_id; datatype=string;  
value=<sip_timer_profile_id>;  
Diversion Indication for SIP Subscribers  
Diversion indication provides supplemental redirection information to the SIP entity receiving a call.  
The SIP entity uses this information to identify from whom the call was diverted, and why the call was  
diverted. It also provides information for each redirection if multiple redirections occurred. This is  
provided in the form of a SIP Diversion header.  
Forwarding information allows applications such as SIP voice-mail servers to access the mailbox of the  
original called party for proper outgoing greeting and message deposit when a forwarded call is received.  
Billing systems also use the information to determine the charged party of the call where it is the last  
forwarding party that is billed.  
The BTS 10200 supports the Diversion Indication feature according to the specifications in the IETF  
document draft-levy-sip-diversion-02.txt. For incoming calls, the BTS 10200 uses the party number  
information from the top-most and bottom-most diversion headers. The BTS 10200 reads the diversion  
count across all diversion headers to determine the total diversion count. For outgoing calls, The  
BTS 10200 sends 0, 1 or 2 diversion headers, depending on the forwarding information of the call.  
Diversion header parameter support is limited to the diversion counter and the diversion reason. These  
two parameters in diversion headers are populated for outgoing calls and interpreted on incoming calls.  
For INVITEs sent out by the BTS 10200, the following behavior applies:  
If no diversion information is available, no diversion headers are included.  
If there is an original called party, one diversion header is added to the outgoing INVITE message.  
If there is a last forwarding party, a second diversion header is added on top of the original called  
party diversion header.  
Each outgoing diversion header is populated with the party number, the diversion reason, and the  
diversion count.  
For Release 5.0, Maintenance Release 1 and later, privacy parameters are sent and received in the  
Diversion header.  
For Release 5.0, Maintenance Release 1 and later, If the original called number (OCN) and/or the  
redirected DN (RDN) are being sent in Diversion headers towards local SIP subscribers, and the  
presentation value is not allowed, the system applies anonymous to them as follows:  
If an OCN exists, it populates the URL as [email protected] in the To header.  
If a Diversion header is added, it populates the user part of the diversion header with  
anonymous.  
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Chapter 2 SIP Subscribers  
Comparison of SIP-Based Features and MGCP-Based Features  
Comparison of SIP-Based Features and MGCP-Based Features  
Table 2-1 lists the MGCP features available (in the Feature column) and then describes how the feature  
differs when it is used as a SIP feature.  
Table 2-1  
MGCP Features and SIP Support  
MGCP-Based Feature  
8XX Toll-Free  
Abbreviation  
8xx  
Support for SIP Phone Compared to Support for MGCP-Based Phone  
Same as MGCP.  
911 Emergency-Service  
911  
Only E911 support (without the suspend procedure for 45 minutes). Basic 911  
with suspend procedure is not supported.  
Emergency Call (911) is supported for SIP endpoints with one caveat: If the  
calling party (SIP subscriber) disconnects the call, the called party control is  
not available. Otherwise, the call will be released. Expanded emergency  
service (E911) does not require this, but basic emergency service (911) does.  
Both 911 and E911 are supported for MGCP endpoints.  
The Public Safety Answering Point (PSAP) is selected based on default user  
location. No mobility is supported.  
Anonymous Call Rejection ACR  
Same as MGCP, when provided by the BTS 10200. Also provided by the  
phone.  
Anonymous Call Rejection ACR_ACT  
Activation  
BTS 10200 functionality is same for SIP subscribers as for MGCP.  
ACR_ACT is also supported on some SIP phones. Depending on the specific  
phone, the feature on the BTS 10200 might work jointly with the feature on the  
phone.  
Anonymous Call Rejection ACR_DEACT BTS 10200 functionality is same for SIP subscribers as for MGCP.  
Deactivation  
ACR_DEACT is also supported on some SIP phones. Depending on the  
specific phone, the feature on the BTS 10200 might work jointly with the  
feature on the phone.  
Automatic Callback  
AC  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Automatic Callback  
Activation  
AC_ACT  
AC_DEACT  
AR  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Automatic Callback  
Deactivation  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Automatic Recall  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Automatic Recall  
Activation  
AR_ACT  
AR_DEACT  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Automatic Recall  
Deactivation  
SIP phone users cannot activate the service. MGCP users cannot activate the  
service toward SIP phone users.  
Busy Line Verification  
CALEA and LI  
BLV  
n/a  
Not supported.  
For information on lawful intercept (LI) and CALEA, see the “Lawful  
Intercept and Enhanced CALEA” chapter in the Cisco BTS 10200 Softswitch  
Network and Subscriber Feature Descriptions.  
Call Block  
CBLK  
Same as MGCP.  
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Chapter 2 SIP Subscribers  
Comparison of SIP-Based Features and MGCP-Based Features  
Table 2-1  
MGCP Features and SIP Support (continued)  
MGCP-Based Feature  
Call Forward Busy 1  
Abbreviation  
CFB  
Support for SIP Phone Compared to Support for MGCP-Based Phone  
Same as MGCP.  
Call Forward Busy  
Variable Activation  
CFBVA  
Single-stage digit collection.  
Call Forward Busy  
Variable Deactivation  
CFBVD  
CFBI  
Same as MGCP.  
Call Forward Busy  
Interrogation  
Single-stage digit collection.  
Call Forward Combined 1 CFC  
Same as MGCP.  
Call Forward Combined  
Activation  
CFC_ACT  
Single-stage digit collection.  
Call Forward Combined  
Deactivation  
CFC_DEACT Same as MGCP.  
Call Forward No Answer 1 CFNA  
Same as MGCP.  
Call Forward No Answer  
Variable Deactivation  
CFNAVA  
Single-stage digit collection.  
Call Forward No Answer  
Variable Deactivation  
CFNAVD  
CFNAI  
CFU  
Same as MGCP.  
Call Forward No Answer  
Interrogation  
Single-stage digit collection.  
Same as MGCP.  
Call Forward  
Unconditional 1  
Call Forward  
Unconditional Activation  
CFUA  
CFUD  
Single-stage digit collection.  
Same as MGCP.  
Call Forward  
Unconditional  
Deactivation  
Call Forward  
Unconditional  
Interrogation  
CFUI  
Single-stage digit collection.  
Call Hold  
CHD  
Functionality provided by the phone. The BTS 10200 supports the interface.  
Call Park  
CPRK  
CPRK_RET  
CT  
Not supported.  
Not supported.  
Call Park and Retrieve  
Call Transfer  
For SIP phones, this feature is provided as part of REFER support on the  
on page 2-31 for details.  
Call Waiting  
CW  
Functionality provided by the phone. The BTS 10200 supports the interface.  
Varies with phone functionality.  
Call Waiting Deluxe  
CWD  
CWDA  
Call Waiting Deluxe  
Activation  
Varies with phone functionality.  
Call Waiting Deluxe  
Deactivation  
CWDD  
Varies with phone functionality.  
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Chapter 2 SIP Subscribers  
Comparison of SIP-Based Features and MGCP-Based Features  
Table 2-1  
MGCP Features and SIP Support (continued)  
MGCP-Based Feature  
Abbreviation  
Support for SIP Phone Compared to Support for MGCP-Based Phone  
Call Waiting Deluxe  
Interrogation  
CWDI  
Varies with phone functionality.  
Calling Identity Delivery  
CIDSD  
CIDSS  
CIDCW  
Presentation status from the phone, and single-stage digit collection.  
Presentation status from the phone, and single-stage digit collection.  
Functionality provided by the phone. Cisco BTS 10200 supports the interface.  
and Suppression (Delivery)  
2
Calling Identity Delivery  
and Suppression  
(Suppression) 2  
Calling Identity Delivery  
on Call Waiting  
Calling Name Delivery 3  
CNAM  
CNAB  
Same as MGCP.  
Calling Name Delivery  
Blocking  
Presentation status from the phone, and single-stage digit collection.  
Calling Number Delivery 3 CND  
The calling party number, if available, is delivered in the From header of the  
outgoing INVITE from the BTS 10200 to the terminating SIP phone. The  
number is delivered to the SIP phone even if the CND feature is not  
provisioned for the subscriber.  
Calling Number Delivery  
Blocking  
CNDB  
Presentation status from the phone, and single-stage digit collection.  
Cancel Call Waiting  
Class of Service  
CCW  
COS  
CDP  
Functionality provided by the phone. Cisco BTS 10200 supports the interface.  
CoS Screening supported, without Auth/Account code collection.  
Custom-Dial-Plan  
Same as MGCP.  
Same as MGCP.  
Not supported.  
Customer Originated Trace COT  
Directed Call Pickup  
without Barge-in  
DPN  
Directed Call Pickup with DPU  
Barge-in  
Not supported.  
Distinctive Alerting Call  
Waiting Indication  
DACWI  
This feature is provided to Centrex users only.  
Provisioning for SIP does not differ from provisioning for MGCP. However,  
the delivery method for DACWI is different.  
The Centrex administrator provisions a list of DNs that are to receive DACWI  
tones.  
In MGCP, the phone plays the tone specified by the BTS 10200 in the protocol  
message. In SIP, the tone provisioned for the DN is specified by the BTS 10200  
in the Alert-Info header of the INVITE as a file URL. A SIP phone, if capable,  
interprets this header and plays the specified distinctive ringing or call-waiting  
tone.  
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Chapter 2 SIP Subscribers  
Comparison of SIP-Based Features and MGCP-Based Features  
Table 2-1  
MGCP Features and SIP Support (continued)  
MGCP-Based Feature  
Abbreviation  
Support for SIP Phone Compared to Support for MGCP-Based Phone  
Distinctive Ringing Call  
Waiting  
DRCW  
Provisioning for SIP does not differ from provisioning for MGCP. However,  
the delivery method for DRCW is different.  
The subscriber provisions a list of DNs to receive DRCW tones.  
In MGCP, the phone plays the tone specified by the Cisco BTS 10200 in the  
protocol message. In SIP, the tone provisioned for the DN is specified by the  
Cisco BTS 10200 in the Alert-Info header of the INVITE as a file URL. A SIP  
phone, if capable, can interpret this header and play the specified distinctive  
ringing or call-waiting tone.  
Distinctive Ringing Call  
Waiting  
DRCW_ACT  
DND  
Same as MGCP.  
Do Not Disturb  
Same as MGCP, except that the reminder ring cannot be used with SIP devices.  
For additional information on DND, see the “Do Not Disturb” section on  
Do Not Disturb Activation DND_ACT  
Same as MGCP.  
Do Not Disturb  
Deactivation  
DND_DEACT Same as MGCP.  
Group Speed Call—1 digit GSC1D  
Group Speed Call—2 digit GSC2D  
Not supported.  
Not supported.  
Not supported.  
Not supported.  
Not supported.  
Not supported.  
Hotline  
HOTLINE  
HOTV  
Hotline Variable  
Hotline Variable Activation HOTVA  
Hotline Variable  
Deactivation  
HOTVD  
HOTVI  
ISFG  
Hotline Variable  
Interrogation  
Not supported.  
Same as MGCP.  
Incoming Simulated  
Facility Group  
Local Number Portability LNP  
Same as MGCP.  
Multiline Hunt Group  
MLHG  
MLHG is not supported for SIP subscribers.  
Multiple Directory Number MDN  
Provisioning for SIP does not differ from provisioning for MGCP. However,  
the delivery methods for distinctive -ringing (a distinctive ring tone for each  
line of the MDN subscriber), and the distinctive tone on call waiting are  
different.  
You provision distinctive ringing and call waiting tones for each DN of the  
MDN subscriber in the same manner for MGCP and SIP. In MGCP, the phone  
plays the tone specified by the Cisco BTS 10200 in the protocol message. In  
SIP, the tone provisioned for the DN is specified by the Cisco BTS 10200 in  
the Alert-Info header of the INVITE as a file URL. A SIP phone, if capable,  
can interpret this header and play the specified distinctive ringing or call  
waiting tone.  
Outgoing Call Barring  
OCB  
Same as MGCP.  
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Chapter 2 SIP Subscribers  
Comparison of SIP-Based Features and MGCP-Based Features  
Table 2-1  
MGCP Features and SIP Support (continued)  
MGCP-Based Feature  
Abbreviation  
Support for SIP Phone Compared to Support for MGCP-Based Phone  
Outgoing Call Barring  
Activation  
OCBA  
Single stage digit collection.  
Outgoing Call Barring  
Deactivation  
OCBD  
OCBI  
OSFG  
Single stage digit collection.  
Single stage digit collection.  
Same as MGCP.  
Outgoing Call Barring  
Interrogation  
Outgoing Simulated  
Facility Group  
Remote Activation of Call RACF  
Forwarding  
Same as MGCP.  
Remote Activation of Call RACF_PIN  
Forwarding PIN  
Same as MGCP.  
Refer  
REFER  
This is not for MGCP users. Cisco BTS 10200 supports the SIP REFER  
interface to enable services such as Call-Transfer (attended, unattended)  
provided by the phone.  
Selective Call Acceptance SCA  
Same as MGCP.  
Same as MGCP.  
Selective Call Acceptance SCA_ACT  
Activation  
Selective Call Forwarding SCF  
Same as MGCP.  
Same as MGCP.  
Selective Call Forwarding SCF_ACT  
Activation  
Selective Call Rejection  
SCR  
Same as MGCP.  
Same as MGCP.  
Selective Call Rejection  
Activation  
SCR_ACT  
Speed Call—1 digit  
SC1D  
Not supported.  
Not supported.  
Speed Call Activation—1 SC1D_ACT  
digit  
Speed Call—2 digit  
SC2D  
Not supported.  
Not supported.  
Speed Call Activation—2 SC2D_ACT  
digit  
Three-Way Calling  
TWC  
Functionality provided by the phone. The BTS 10200 supports the interface.  
Varies with phone functionality.  
Three-Way Call Deluxe  
TWCD  
USTWC  
Usage-Sensitive  
Functionality provided by the phone. The BTS 10200 supports the interface.  
Three-Way Calling  
Warmline  
WARMLINE  
Not supported.  
1. See additional information on call forwarding features in the “Call Forwarding” section on page 2-19.  
2. See additional information on the delivery and suppression feature in the “Caller ID Delivery Suppression” section on page 2-21.  
3. See additional information on calling name and calling number in the “Calling Name and Number Delivery” section on page 2-20.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Cisco BTS 10200 Softswitch-Based Features  
Softswitch-based features are directly provided by the BTS 10200. SIP phones can provide some  
features on their own; for information on the features provided by the different SIP phones, see the SIP  
phone administration guides.  
This section describes Softswitch-based features entirely provided by the BTS 10200.  
Note  
BTS 10200 announcements are customizable on a business group basis. If an announcement is not  
provisioned or cannot be played, a reorder tone is played.  
Summary  
Table 2-2 lists the most commonly used Softswitch-based features; however, it is not an exhaustive list.  
The sections that follow the table provide additional details on selected softswitch-based features.  
Note  
s
Table 2-2  
BTS 10200-Based SIP Features  
SIP Feature  
Acronym  
ACR  
Activation and Deactivation of Anonymous Call Rejection  
Anonymous Call Rejection Activation  
Anonymous Call Rejection Deactivation  
Call Forwarding  
ACR_ACT  
ACR_DEACT  
CF  
Call Forwarding on Busy Variable Activation  
Call Forwarding on Busy Variable Deactivation  
Call Forwarding on Busy Interrogation  
Call Forwarding on No Answer Variable Activation  
Call Forwarding on No Answer Variable Deactivation  
Call Forwarding on No Answer Interrogation  
Call Forwarding Unconditional Activation  
Call Forwarding Unconditional Deactivation  
Call Forwarding Unconditional Interrogation  
Call Waiting Deluxe Activation  
CFBVA  
CFBVD  
CFBI  
CFNAVA  
CFNAVD  
CFNAI  
CFUA  
CFUD  
CFUI  
CWDA  
CWDD  
CWDI  
Call Waiting Deluxe Deactivation  
Call Waiting Deluxe Interrogation  
Called Party Termination  
CPT  
Caller ID Suppression  
CIDS  
Calling Identity Delivery and Suppression (per call)—Suppression part CIDSS  
Calling Identity Delivery and Suppression (per call)—Delivery part  
Calling Name Delivery Blocking  
CIDSD  
CNAB  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Table 2-2  
SIP Feature  
BTS 10200-Based SIP Features (continued)  
Acronym  
CND  
Calling Name and Number Delivery  
Customer Access Treatment  
CAT  
Customer-Originated Trace  
COT  
Differentiated Services Code Point  
Direct Inward Dialing  
DSCP  
DID  
Direct Outward Dialing  
DOD  
Do Not Disturb  
DND  
Do Not Disturb Activation  
DND_ACT  
DND_DEACT  
E911  
Do Not Disturb Deactivation  
Emergency Call  
E.164 and Centrex Dialing Plan (Extension Dialing)  
Incoming and Outgoing Simulated Facility Group  
Multiple Directory Numbers  
Operator Services (0-, 0+, 01+, 00 calls)  
Outgoing Call Barring  
E.164  
ISFG and OSFG  
MDN  
OCB  
Outgoing Call Barring Activation  
Outgoing Call Barring Deactivation  
Outgoing Call Barring Interrogation  
Remote Activation of Call Forwarding  
Vertical Service Codes  
OCBA  
OCBD  
OCBI  
RACF  
VSC  
Call Forwarding  
The differences between the feature for SIP and the feature for MGCP are as follows:  
There is no tone provided for SIP users to prompt for forwarding digits. The SIP users enter the  
forwarding digits immediately after the VSC. This is called single-stage dialing.  
There is no dial tone played after the SIP user successfully activates or deactivates the Forwarding  
features. The SIP user always hears an announcement (if announcements are provisioned) or a  
re-order tone.  
Call Forwarding Activation and Deactivation  
Activation and deactivation of call forwarding features use the vertical service code (VSC), also known  
as a star code.  
With SIP support, the call forwarded to number can be a Centrex extension number (only applicable for  
business users) or an E.164 number.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Note  
Forwarding to a URL address of record (AOR) is not supported.  
SIP subscribers do not hear a final dial tone upon completing activation or deactivation. Instead, an  
announcement plays for the subscriber, indicating that the status of the forwarding feature is being  
activated or deactivated. This is irrespective of the Final Stage Dial Tone (FDT) flag (Y/N) provisioned  
for these features.  
Call Forwarding to an E.164 Number or an Extension Number  
Activation and deactivation are accomplished using single-stage dialing.  
Detailed Provisioning Procedure and Feature Description  
Additional information on this feature is provided at the following links.  
Feature Behavior (Feature Description and  
Handset Provisioning)  
Provisioning Procedure  
Call forwarding sections in the  
Cisco BTS 10200 Softswitch Provisioning Guide Cisco BTS 10200 Softswitch Network and  
Subscriber Feature Descriptions document  
Calling Name and Number Delivery  
Calling number delivery (CND) provides the SIP subscriber endpoint with the calling number of an  
incoming call. Calling name delivery (CNAM) provides the endpoint with the name of the calling party.  
CND  
The calling party number, if available, is delivered in the From header of the outgoing INVITE from the  
BTS 10200 to the terminating SIP phone. The number is delivered to the SIP phone even if the CND  
feature is not provisioned for the subscriber. The delivered information is as follows:  
If the calling number is available and the presentation indication is not restricted, the number is  
inserted into the user information portion of the From header.  
If the calling number is available and the presentation indication is restricted, the user information  
portion of the From header is set as “Anonymous.”  
If the calling number is not available, the user information portion of the From header is left empty.  
CNAM  
The calling party name is delivered in the outgoing INVITE from the BTS 10200 to the terminating SIP  
phone only if the CNAM feature is provisioned for the SIP subscriber. The delivered information is as  
follows:  
If the calling number and name are available and the presentation indication of both the calling  
number and calling name are not restricted, the calling name is inserted into the display name field  
of the From header.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
If the calling number and name are available and the presentation indication of either calling number  
or calling name is restricted, the display name field of the From header is set as “Anonymous.”  
If the calling name is not available, the display name field of the From header is left empty.  
Additional information on this feature is provided at the following links.  
Feature Behavior (Feature Description and  
Handset Provisioning)  
Provisioning Procedure  
CND, CNAM, CNDB, and CNAB sections in the “Calling Identity Features” section in the  
Cisco BTS 10200 Softswitch Provisioning Guide Cisco BTS 10200 Softswitch Network and  
Subscriber Feature Descriptions document  
Caller ID Delivery Suppression  
The treatment for caller’s identity is based on the presence of “anonymous” in the Display-Name field  
of the From header in the INVITE message. If the caller’s identity is restricted in the incoming SIP  
INVITE message, the presentation is suppressed.  
Caller Identity presentation (allowed/restricted) information for SIP subscribers is not maintained in the  
the BTS 10200 database. This information is maintained on the individual phones and can be  
provisioned through the phone softkeys. Permanent restriction on the phone can be overridden if the  
caller dials a feature (*) code on a per-call basis. This is a single-stage dialing for SIP subscribers.  
Additional information on this feature is provided at the following links.  
Feature Behavior (Feature Description and  
Handset Provisioning)  
Provisioning Procedure  
“CND, CNAM, CNDB, and CNAB” sections in  
(CIDSD and CIDSS)” section in the  
Cisco BTS 10200 Softswitch Network and  
Subscriber Feature Descriptions document  
Customer Access Treatment  
Provisioning this feature for SIP is the same as provisioning it for MGCP. The provisioning commands  
for this feature are shown in Chapter 8, “Centrex Provisioning,” in the Cisco BTS 10200 Softswitch  
Provisioning Guide.  
Direct Inward Dialing  
Provisioning this feature for SIP is the same as provisioning it for MGCP.  
Assign the DID number to the subscriber as DN1 in the Subscriber table.  
For information about the operation of this feature, see the “Direct Inward Dialing” section in the  
Cisco BTS 10200 Softswitch Network and Subscriber Feature Descriptions guide.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Direct Outward Dialing  
With the Direct Outward Dialing (DOD) service, a station user can place external calls to the exchange  
network without attendant assistance by:  
1. Dialing the DOD (public) access code (usually the digit 9)  
2. Receiving a second dial tone  
3. Dialing the external number (a number outside the customer group)  
Access to the DOD feature is subject to station restrictions.  
Note  
For IP phones, the second dial tone is provided by the phone itself. However, the prefix code is presented  
to the BTS 10200 along with the DDD number in the INVITE message. Secondary dial-tone capability  
is dependent on the SIP device used.  
For information about the operation of this feature, see the DOD for PBX section in the  
Cisco BTS 10200 Softswitch Network and Subscriber Feature Descriptions guide.  
Do Not Disturb  
The Do Not Disturb (DND) feature enables a user to block incoming calls to the station on which the  
feature is activated. If no call forwarding features are activated, calls to the station are routed to busy  
treatment. This feature should be provisioned and activated on the BTS 10200 because of feature  
interaction with advanced features like executive override.  
This is a single-stage dialing activation feature. The Alert-Info header plays the result of  
activation/deactivation—Success is a confirmation tone and failure is a failure message.  
The reminder ring option (which is available with the DND feature on MGCP-based lines) cannot be  
used with SIP devices.  
For features (such as DND) that can be fully provisioned on the BTS 10200 or on the phone, you can  
provision either one of the devices to enable the feature.  
Caution  
Prior to provisioning your system, determine how you want to apply and configure features in your  
network to avoid conflicts between features provided by the BTS 10200 and features provided by the  
phones.  
Additional information on this feature is provided at the following links.  
Feature Behavior (Feature Description and  
Handset Provisioning)  
Provisioning Procedure  
Cisco BTS 10200 Softswitch Provisioning Guide Cisco BTS 10200 Softswitch Network and  
Subscriber Feature Descriptions document  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
E.164 and Centrex Dialing Plan (Extension Dialing)  
The system supports E.164 and Centrex Dialing Plan (extension dialing) addressing from SIP  
subscribers served by the local BTS 10200.  
The SIP phone’s dial plan must be configured so that it considers the number of digits in the Centrex  
group. Centrex dialing can be provisioned within a range of 1 through 7 digits. Each Centrex group  
should have its own separate dial plan.  
Note  
The CDP feature should be assigned to every Centrex category user.  
Example 2-1 A SIP URL with E.164 Addressing  
sip:[email protected];user=phoneA sip:[email protected];user=phone  
Additional information on this feature is provided at the following links.  
Feature Behavior (Feature Description and  
Handset Provisioning)  
Provisioning Procedure  
Cisco BTS 10200 Softswitch Provisioning Guide section in the Cisco BTS 10200 Softswitch  
Network and Subscriber Feature Descriptions  
document  
in the Cisco BTS 10200 Softswitch Network and  
Subscriber Feature Descriptions document  
Operator Services (0-, 0+, 01+, and 00 Calls)  
There is no Cisco BTS 10200 Softswitch subscriber-specific provisioning involved for Operator  
Services.  
Additional information on this feature is provided in the “Operator Services” section in the  
Cisco BTS 10200 Softswitch Network and Subscriber Feature Descriptions document  
User-Level Privacy  
User-level privacy is provisioned in the Subscriber table.  
Setting the privacy parameter to user directs the system to apply the user-provided privacy information.  
This setting (privacy=user) applies only to SIP endpoints that are capable of including privacy  
information.  
Vertical Service Code Features  
This section explains how to plan vertical service codes (VSCs) in a network with SIP subscribers, and  
lists the VSC-enabled features.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Planning VSCs In Networks with SIP Subscribers.  
Some features require SIP subscriber to enter a series of numbers and characters on the SIP client or  
handset. Typically, the subscriber dials VSC digits followed by additional dialing keys representing the  
parameters for the feature call. For MGCP subscribers, the BTS 10200 sends a response tone or  
announcement between the VSC code and the additional digits. However, for SIP endpoints, all the digits  
are dialed at a stretch without waiting for an intervening response tone from the BTS 10200. The  
following paragraph explains how certain combinations of VSC can cause mismatches between the  
feature the subscriber is attempting to manage versus the response of the BTS 10200, and how to plan  
VSCs to avoid these mismatches.  
You should not deploy certain combinations of VSCs on networks with SIP endpoints. If you deploy a  
VSC longer than 2 digits, make sure that the longer VSC does not begin with the same sequence of  
characters as one of the shorter VSCs. In some cases, the system might match the shorter string even if  
the subscriber dialed the longer string. Consider the following example, for which the subscriber is  
expected to dial a VSC followed by a DN.  
A SIP subscriber is provisioned with *93 for Feature1 and *938 for Feature2, and dials  
*938+2135551801 to invoke Feature2. The BTS 10200 receives *9382135551801 in the INVITE  
message. By default, it takes the first six characters, in this case *93821, and uses this string to look up  
the feature in the VSC table. There is no match for *93821, therefore the BTS 10200 proceeds as follows.  
First, it uses *9 to look for a match in the VSC table and it cannot be found. Then it uses *93, finds a  
match, and delivers Feature1. This is incorrect. The user's intention was to invoke Feature2 and not  
Feature1. The solution is for the service provider to change one of the two VSCs (either *93 or *938) in  
the VSC table.  
Supported VSC-Enabled Features for SIP Endpoints  
The following BTS 10200 Vertical Service Code (VSC) features are supported on SIP endpoints:  
Calling identity delivery and suppression, suppression part (CIDSS)  
Calling identity delivery and suppression, delivery part (CIDSD)  
Calling name delivery blocking (CNAB)  
Outgoing call bearing activation (OCBA), outgoing call bearing deactivation (OCBD), outgoing call  
bearing interrogation (OCBI)  
Call forwarding unconditional activation (CFUA), call forwarding unconditional deactivation  
(CFUD), call forwarding unconditional interrogation (CFUI)  
Reminder ringback cannot be enabled for SIP subscribers. If you are turning on the Call Forward  
Unconditional (CFU) feature for a SIP subscriber, make sure that reminder ring capability is turned  
off. This should be done at a subscriber level.  
Here is the command format at the feature level:  
add feature fname=CFU; tdp1=TERMINATION_ATTEMPT_AUTHORIZED;  
tid1=TERMINATION_ATTEMPT_AUTHORIZED; feature_server_id=FSPTC235; ttype1=R;  
fname1=CFUA; fname2=CFUD; type1=MCF; value1=Y; type2=RR; value2=N;  
description=CFU MCF multiple call forwarding allowed, RR ring reminder not  
allowed;  
And at the subscriber feature level:  
add subscriber-feature-data sub_id=sip_sub2; FNAME=CFU; type2=RR; VALUE2=N;  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Call forwarding on no answer variable activation (CFNAVA), call forwarding on no answer variable  
deactivation (CFNAVD), call forwarding on no answer interrogation (CFNAI)  
Call forwarding on busy variable activation (CFBVA), call forwarding on busy variable deactivation  
(CFBVD), call forwarding on busy variable interrogation (CFBI)  
RACF Pin Change  
Voice Mail  
The voice-mail (VM) feature on the BTS 10200 allows subscribers to retrieve waiting voice messages  
from a VM server. The BTS 10200 receives a message-waiting indication (MWI) from the VM server  
and forwards the MWI to the subscriber’s handset. The subscriber can then retrieve messages from the  
server. The VM feature is available to individual subscribers and Centrex subscribers.  
SIP trunks interconnecting the BTS 10200 to an external VM server must be provisioned as SIP VM  
trunks. To do that, you set the VM flag (voice-mail-trunk-grp) for these trunks in the softsw-tg-profile  
table. (See the “SIP Trunk to Voice-Mail Server” section on page 3-46.)  
Note  
For a description of the basic VM feature, see the Voice Mail and Voice Mail Always” section in the  
Cisco BTS 10200 Softswitch Network and Subscriber Feature Descriptions document. For general VM  
provisioning details, see the “Provisioning Voice Mail” section in the Cisco BTS 10200 Softswitch  
Provisioning Guide.  
VM Actions  
The following voice mail-related actions are supported in the BTS 10200:  
VM Deposit  
There are two methods for depositing voice mail. In the first, the subscriber dials the pilot number for  
the VM server, and the call terminates on the voice-mail trunk. The VM system then collects the message  
for a target mailbox, using Interactive Voice Response (IVR) prompts to guide the subscriber.  
This method of depositing voice mail does not use any special BTS 10200 capabilities; it just requires  
that the VM SIP trunk is provisioned and the pilot number is added to the dial plan of the subscriber  
calling the VM system.  
In the second (more common) method, the subscriber activates a call forwarding feature on the  
BTS 10200, such as CFNA, CFU, or CFB, and specifies the forwarding number as the pilot number of  
the VM server.  
MWI Notification  
When a SIP phone registers with the BTS 10200, the BTS 10200 sends an unsolicited SIP NOTIFY  
message to convey the MWI status to the phone. This occurs on every registration, including refreshes.  
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Cisco BTS 10200 Softswitch-Based Features  
Whenever a change in VM status occurs for a subscriber (for example, when a VM message is deposited  
for the subscriber, or when all such messages have been retrieved), the VM server sends an update to the  
BTS 10200. If the subscriber is on a SIP phone, the BTS 10200 sends an unsolicited SIP Notify message  
to convey the MWI status to the phone. The number in the Notify message Request URL (which is the  
assigned subscriber number) identifies the subscriber.  
When the BTS 10200 is congested by a flood of registrations (which might occur, for example, when  
power is restored to a region after an outage), it can automatically suppress the MWI indication to the  
registering phones, so that registration throughput is not adversely affected.  
The BTS 10200 implements the draft-ietf-sipping-MWI-01.txt with the following caveat: It supports  
receiving unsolicited NOTIFYs from a VM system; however, it does not support subscribing to these  
notifications. Further, the BTS 10200 does not support subscriptions for MWI. It sends unsolicited  
NOTIFYs for MWI to SIP subscribers. No subscription is expected from the SIP phones for the purpose  
of receiving this notification.  
The notification of MWI by the BTS 10200 is enabled by default (VMWI=Y in the Subscriber table).  
You can disable it by setting VMWI=N.  
Tip  
For MGCP subscribers, the BTS 10200 sends the MGCP RQNT message to turn on MWI on the analog  
phone. This activates the MWI indicator on the subscriber phone. The indicator can be visual (a lamp,  
an envelope, or another icon on a display) or it can be auditory, such as a stutter dial tone that is provided  
when the user next goes off-hook.  
For information on setting the MWI and VMWI parameters in the Subscriber table, see the “Message  
Waiting Indicator (MWI)—Audible and Visual” section in the Cisco BTS 10200 Softswitch Network and  
Feature Descriptions document.  
Retrieving VM  
To retrieve a VM message, subscribers dial the pilot number for the VM server. The BTS 10200 routes  
the call to the SIP trunk for VM, based on the provisioned dial plan for the subscriber and the route,  
destination, and trunk-group entries.  
Once the VM message is retrieved, the VM server sends a Notify message to the BTS 10200 to turn off  
the MWI indicator.  
Calling Back a Message Depositor  
When subscribers call into a VM server, this feature allows for calling back the person who left the  
voice-mail message. The feature requires that a Softswitch trunk for the VM server be provisioned in the  
Cisco BTS 10200 Softswitch with the relevant routes, destination, and dial plans in order to admit  
VM-originated calls into the BTS 10200.  
VM Implementation for Centrex Subscribers  
For calls received on SIP VM trunks from the VM server, a subscriber is provisioned and associated as  
the main sub-ID for each trunk. The subscriber information represents properties of a specific Centrex  
group and does not represent any particular subscriber. No AOR is provisioned for this subscriber. This  
information is used for call processing.  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Note  
For general VM provisioning details, see “Provisioning Voice Mail” in the Cisco BTS 10200 Softswitch  
Provisioning Guide.  
VM Within a Single Centrex Group  
The following examples show commands for provisioning Centrex VM. Before you perform the  
following steps, you must already have a Centrex group provisioned on your system. See the procedure  
in the “Provisioning a Centrex Group” section of the Cisco BTS 10200 Softswitch Provisioning Guide.  
Step 1  
Step 2  
Add the destination ID for the voice-mail main subscriber.  
add destination dest-id=tb16-local; call-type=LOCAL; route-type=SUB;  
Add a dial plan profile and dial plan for a SIP trunk to the VM server.  
add dial-plan-profile id=tb16;  
add dial-plan id=tb16; digit-string=469-555; dest-id=tb16-local; min-digits=10;  
max-digits=10  
Step 3  
Add the softswitch trunk group profile for voice mail.  
add softsw-tg-profile id=VM_Profile; protocol-type=SIP; voice_mail_trunk_grp=Y;  
Note  
As an option, you can provision the diversion-header-supp token in the softsw-tg-profile table  
to Y. This instructs the VM server to select the target inbox based on the original called number  
in the Diversion header of the SIP message.  
Step 4  
Add the SIP trunk group.  
Note  
This SIP trunk group serves several purposes. It is used (1) by the subscriber to access the VM  
server, (2) by the BTS 10200 to forward incoming calls to the VM server, and (3) by the VM  
server to notify the BTS 10200 that a message is waiting for the subscriber.  
add trunk-grp id=80032; softsw-tsap-addr=vm.domainname.com:5060;  
call-agent-id=CA146; tg-type=softsw; tg-profile-id=VM_Profile; dial-plan-id=tb16  
Step 5  
Add a subscriber to the Centrex group to serve as the VM main subscriber.  
add subscriber id=vmctxg1; CATEGORY=ctxg; BILLING-DN=469-555-4444;  
DN1=469-555-4444; SUB-PROFILE-ID=Centrex_sp2; TERM-TYPE=TG; ctxg_id=ctxgsip1;  
tgn_id=80032;  
Step 6  
Step 7  
Link the VM main subscriber with the trunk group.  
change trunk-grp; id=80032; main_sub_id=vmctxg1;  
Map the voice-mail Centrex extension to the VM main subscriber.  
add ext2subscriber CTXG-ID=ctxgsip1; EXT=54444; CAT-CODE=1; SUB-ID=vmctxg1;  
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Chapter 2 SIP Subscribers  
Cisco BTS 10200 Softswitch-Based Features  
Step 8  
If your VM server does not support FQDN hostnames, you must provision a serving-domain-name  
record in the BTS 10200 using the IP addresses resolved from the sia-xxxCAnnn.domain address.  
Otherwise, the VMWI status from SIP voice-mail platforms fails authentication with the BTS 10200.  
The details for this step are provided in Step 6 of the “SIP Trunk to Voice-Mail Server” section on  
page 3-46,  
Provisioning Voice Mail Across Multiple Centrex Groups  
A VM application server can provide VM service for Centrex subscribers from multiple Centrex groups  
on the BTS 10200. For the VM server to identify the subscriber and provide service configured for a  
Centrex group, the BTS 10200 must indicate the Centrex group with which the subscriber is associated,  
When the BTS 10200 forwards a call from a Centrex extension to VM, the VM server identifies the  
Centrex group of the extension to deposit the message in the correct mailbox. Further, when the VM  
server sends a SIP Notify message to indicate that messages are waiting for a Centrex subscriber on the  
BTS 10200, it must identify the Centrex group in the request URI of the NOTIFY message sent to the  
BTS 10200.  
For any INVITE sent out a SIP trunk by the BTS 10200 to the VM server, a BTS 10200 proprietary SIP  
URL parameter bgid is added to the From, To, Diversion, and Request URIs, if the user part of those  
URLs contains a Centrex extension number format in the user information field. The bgid value is  
provisioned as the trunk-subgroup-type on the SIP trunk, and identifies the Centrex group.  
An example of this parameter syntax follows:  
INVITE sip:[email protected]:5060;user=phone;bgid=grpA SIP/2.0  
From: <sip:[email protected];user=phone;bgid=grpA>;tag=1_1146_f40077_3jwv  
To: <sip:[email protected];user=phone;bgid=grpA>  
Diversion: <sip:[email protected];bgid=grpA>;reason=unconditional;counter=1  
When the VM server notifies the BTS 10200 of a MWI for a Centrex subscriber, the VM server sends a  
Notify SIP request to the BTS 10200 with a Centrex number format in the Request URL, and an  
associated bgid parameter identifying the Centrex group associated with the subscriber. When the VM  
server initiates a call to a BTS 10200 Centrex subscriber for VM callback functionality, bgid is added to  
the request URL of the initial INVITE originating from the VM server. This identifies the Centrex group  
associated with the subscriber.  
The BGID parameter in the ReqUri of an INVITE originated from the VM server identifies the called  
subscriber in the targeted Centrex group. For example, the BGID parameter in the ReqUri of a NOTIFY  
message from the VM server to the BTS 10200 identifies the subscriber in the targeted Centrex group  
whose MWI lamp is turned on or off.  
The BTS 10200 does not support extension-dialed calls from one Centrex group to another. Therefore,  
the bgid parameter has an identical value if it is present in any of the URLs in the From, To, Diversion,  
and Request URL headers for a given INVITE message. The trunk group configuration includes a trunk  
subgroup field for specifying the bgid parameter value. One trunk group is provisioned for each Centrex  
group; the bgid parameter in the trunk group table is unique to the specific Centrex group. Routing tables  
are configured so that each trunk handles SIP calls to and from the VM server for a specific Centrex  
group. To qualify a specific trunk for bgid and VM, provision as follows:  
In the Trunk Group (trunk-grp) table, provision the bgid value in the trunk-sub-grp field.  
In the Softswitch Trunk Group Profile (softsw-tg-profile) table:  
Provision the trunk-sub-grp-type field as BGID.  
Provision the voice-mail-trunk-grp field as Y.  
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Chapter 2 SIP Subscribers  
Jointly Provided Features  
The following provisioning steps illustrate how to provide VM service for BTS 10200 Centrex  
subscribers across multiple Centrex groups.  
Step 1  
Step 2  
Add a SIP trunk profile for voice-mail trunks. Qualify voice-mail trunks by setting the voice-mail flag,  
and set the trunk sub-group type to indicate use of business group identifier:  
add softsw_tg_profile ID=<profile_id>; PROTOCOL_TYPE=SIP;  
VOICE_MAIL_TRUNK_GRP=Y; TRUNK_SUB_GRP_TYPE=BGID;  
Add a SIP trunk for each business group identifier. Each trunk points to the address of the voice-mail  
sever.  
In the following command, be sure to enter a unique business group identifier for each Centrex group,  
for example, bg1, bg2, and bg3, for the three Centrex groups in this example.  
Also, be sure to specify the FQDN and port that the VM server uses for SIP message exchange, for  
example, vmserver:5060.  
add trunk_grp ID=<trk_grp_id1>; TG_TYPE=SOFTSW; TG_PROFILE_ID=<profile_id>;  
SOFTSW_TSAP_ADDR=vmserver:5060; DIAL_PLAN_ID=dp; TRUNK_SUB_GRP=bg1;  
add trunk_grp ID=<trk_grp_id2>; TG_TYPE=SOFTSW; TG_PROFILE_ID=<profile_id>;  
SOFTSW_TSAP_ADDR=vmserver:5060; DIAL_PLAN_ID=dp; TRUNK_SUB_GRP=bg2;  
add trunk_grp ID=<trk_grp_id3>; TG_TYPE=SOFTSW; TG_PROFILE_ID=<profile_id>;  
SOFTSW_TSAP_ADDR=vmserver:5060; DIAL_PLAN_ID=dp; TRUNK_SUB_GRP=bg3;  
Step 3  
Create a dial plan for calls received on the SIP trunks, so that they can be routed based on the called party  
number. For example, the identifier for the dial plan in this example is dp. (The dial plan provisioning  
details are not shown here.) The Centrex group routing and dial plan tables should be provisioned so that  
calls originating from a specific Centrex group subscriber are sent out the SIP trunk with the business  
group identifier representing that Centrex group.  
Jointly Provided Features  
Some features are provided jointly by the phone and by the BTS 10200. Here are some examples:  
The sections that follow provide information about these features.  
Text-GUI Features  
The BTS 10200 supports SIP client/handset text-based user interface (UI) provisioning for a select set  
of features. This is in addition to numerous supplementary features supported natively by the SIP  
client/handset itself. Some of the features require updating the status on the network database.  
Cisco BTS 10200 provides complementary support to SIP clients/handsets to update end user feature  
access status on the switch network database.  
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Jointly Provided Features  
Provisioning in this context refers to feature activation or deactivation, and setting any applicable  
directory numbers (DNs) associated with the feature. If a SIP handset is used, the phone’s LCD panel is  
used as a menu display area to guide the user toward feature provisioning. If a SIP software client is used,  
the UI display region in the client software is used to guide the user through feature provisioning.  
There might be multiple lines on the SIP phone, but currently services configured by softkeys on the  
phone are available to only one of those lines. The subscriber for that line is provisioned in the  
BTS 10200 with the MAC address of the phone (see the MAC Address to Subscriber table in the  
Supported Handsets  
Cisco BTS 10200 supports any SIP client/handset that supports CallManager XML 3.0.  
Supported Features  
The following features have SIP client/handset-based provisioning support:  
Call Forwarding Unconditional Activation (CFUA), Call Forwarding Unconditional Deactivation  
(CFUD)  
Call Forwarding on Busy Variable Activation (CFBVA), Call Forwarding on Busy Variable  
Deactivation (CFBVD)  
Call Forwarding on No Answer variable Activation (CFNAVA), Call Forwarding on No Answer  
Variable Deactivation (CFNAVD)  
Do Not Disturb Activation (DND-ACT), Do Not Disturb Deactivation (DND-DEACT)  
Anonymous Call Rejection Activation (ACR-ACT), Anonymous Call Rejection Deactivation  
(ACR-DEACT)  
Accessing Features  
The following sections describe how to access the features.  
SIP Handset  
The SIP handset provides a button labeled “Services” or an icon indicating “Services.” Initial access to  
feature provisioning is through the Services button. After initial access, the UI display area provides a  
menu-driven interface and a feature-specific menu.  
To navigate the menu, the end user presses the Up and Down arrow buttons or menu numbers. At any  
level of navigation, the end user presses the Exit softkey to go back one step in the menu hierarchy. The  
user selects menu items using the Select softkey button and uses the numeric dial to enter DN  
information.  
Menu Hierarchy  
Feature Options  
Call Forwarding  
Call-Fwd Busy  
Activate/Deactivate Feature  
Set/Change Forwarding Number  
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Chapter 2 SIP Subscribers  
Jointly Provided Features  
Number:  
Call-Fwd Unconditional  
Activate/Deactivate Feature  
Set/Change Forwarding Number  
Number:  
Call-Fwd No Answer  
Activate/Deactivate Feature  
Set/Change Forwarding Number  
Number:  
Anonymous Call Rejection  
Activate/Deactivate Feature  
Do Not Disturb  
Activate/Deactivate Feature  
SIP Software Clients  
The user interface for applicable software clients is similar to a SIP handset user interface.  
Call Transfer (Blind and Attended) with REFER  
The SIP Call Transfer (CT) feature is supported for SIP subscribers. For SIP phones, this feature is  
provided as part of REFER support on the BTS 10200.  
The CT feature requires phone support for sending the SIP REFER message. See the phone  
documentation for details on the user interface and procedures for effecting a call transfer. Both blind  
and attended transfers are supported. Attended transfer to a transfer-target is supported only after the  
target answers; that is, consultative attended transfer is supported. Attended transfer is not possible while  
the transfer-target is being alerted (ringing state).  
The difference between provisioning the feature for SIP and provisioning it for MGCP is as follows:  
Call transfer on both the Cisco IP Phone 7905/7912 and the Cisco IP Phone 7940/7960 is done using  
softkeys. On the Cisco ATA 186/188, call transfer is done using the Flash key (or by pressing the  
on-hook button briefly) on the analog phone attached to the Cisco ATA 186/188.  
Call-transfer functionality for SIP-based systems is performed using the REFER feature, not the  
traditional call transfer (CT) feature. To enable CT for SIP subscribers, you must provision the  
REFER feature as an office trigger in the Cisco BTS 10200 Softswitch. See the “SIP Call Transfer  
with REFER and SIP INVITE with Replaces” section on page 3-40 for additional details and  
provisioning procedures.  
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Chapter 2 SIP Subscribers  
Phone-Based Features  
Distinctive Ringing  
Distinctive ringing uses a special ringing pattern to alert the called user of incoming calls from  
preselected telephone numbers. This is a CLASS feature and is offered to both business and residential  
users. There is no difference between provisioning the feature for SIP and provisioning it for MGCP.  
You can edit the list of selected numbers though the Screening List Editing (SLE) feature, which requires  
the configuring of an IVR with the BTS 10200. Distinctive ringing can be assigned to a station and to  
the group, and it can be applied to users based on the call type/calling number. When assigned to a group,  
distinctive ringing is applied to users in the group based on the call type. When assigned to the line,  
distinctive ringing is applied to the user based on the calling number. The BTS 10200 sends an Alert-Info  
header in the outgoing INVITE message, instructing the SIP phone to play a specific ring tone.  
Distinctive ringing depends on the SIP phone’s capability to support processing of the information  
received in an Alert-Info header.  
Distinctive Ringing for Centrex DID Calls  
The BTS 10200 sends an Alert-Info header in the outgoing INVITE message, instructing the SIP phone  
to play a specific ring tone. Distinctive ringing depends on the SIP phone’s capability to process the  
information received in the Alert-Info header. There are no differences between provisioning the feature  
for SIP and provisioning it for MGCP.  
Phone-Based Features  
The phone provides some features standalone, without BTS 10200 support. If the SIP phone requires  
provisioning to provide this function, refer to the SIP phone documentation for instructions.  
Table 2-3 lists the phone-based features.  
.
Table 2-3  
SIP Phone-Based Features  
Feature  
Acronym  
CHD  
Call Hold and Resume  
Call Waiting  
CW  
Call Waiting Caller ID  
Cancel Call Waiting  
CODEC Up-Speeding  
Do Not Disturb  
CWCID  
CCW  
CODEC1  
DND  
Three-Way Calling  
TWC  
1. For feature calls between MGCP and SIP subscribers, the BTS 10200 supports the CODEC  
up-speeding capability. The SIP phone would also need to support this capability for the up-speeding  
capability to be fully supported in the call.  
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Phone-Based Features  
For features (such as DND) that are available independently on the phones and the BTS 10200, you can  
provision either device to enable the feature.  
Caution  
Prior to provisioning your system, determine how you want to apply and configure features in your  
network to avoid conflicts between features provided by the BTS 10200 and features provided by the  
phones.  
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Phone-Based Features  
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