Table Of Contents
Provisioning SIP Devices
Provisioning SIP Devices
Configuring a Cisco ATA 186/188 Device
Configuring Cisco ATA 186/188 Adaptor-Specific Required Information
Configuring a Cisco IP Phone 7905
Configuring Cisco IP 7905 Phone-Specific Required Information
Provisioning the Cisco IP Phone 7960 for Initial Setup
Creating a Cisco 7960 Phone-Specific Configuration
Connect Cisco IP Phone 7960 to Cisco BTS 10200
Cisco BTS 10200 Softswitch Phone Mapping
Services Key: Enabling Feature Activation or Deactivation
GUI Feature Server Provisioning
Configuration
Office Provisioning
SIP Subscriber Services
MAC to Subscriber
Setting Up Services
Provisioning a SIP Subscriber
Provisioning Subscriber Features
Activation and Deactivation of Anonymous Call Rejection
Billing
CALEA Call Content
Call Forwarding
Call Forwarding to an E.164 Number or an Extension Number
Calling Name and Number Delivery
Caller ID Delivery Suppression
Called Party Termination
Cisco BTS 10200 Supplementary Vertical Service Code Features
Customer Access Treatment
Customer-Originated Trace
Direct Inward Dialing
Direct Outward Dialing
Office Provisioning
Do Not Disturb
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
Remote Activation of Call Forwarding
Type of Service
Provisioning Network-Level ToS
Provisioning Type of Service Default Settings for SIP Subscribers
Phone-Based Features
Jointly-Provided Features
Session Timer
Call Transfer (Blind and Attended) via Refer Feature
Distinctive Ringing
Distinctive Ringing for Centrex DID Calls
Provisioning SIP Devices
The purpose of this chapter is to serve as a basic guidance for configuring Cisco SIP devices, including:
•
Cisco ATA 186/188
•
Cisco IP Phone 7905
•
Cisco IP Phone 7912
•
Cisco IP Phone 7940
•
Cisco IP Phone 7960
The chapter also demonstrates how to provision SIP subscribers on Cisco devices in to the Cisco BTS 10200 Softswitch, and provides guidance on provisioning and enabling features for SIP subscribers in Cisco BTS 10200.
You can find the detailed step-by-step administration guide for the Cisco ATA 186/188 adaptors at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/index.htm
You can find the detailed step-by-step administration guide for the Cisco 7905/7912 phones at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7905g/addprot/index.htm
You can find the detailed step-by-step administration guide for the Cisco 7940/7960 phones at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sadmin31/index.htm
For multiple line SIP phones, each line must be provisioned with a DN/Subscriber entry in the Cisco BTS 10200 Softswitch.
For information on the supported SIP protocol features, refer to the Cisco BTS 10200 Softswitch Release 4.4 SIP Protocol Support User Guide.
Provisioning SIP Devices
Cisco IP phones are full-featured telephones that can be plugged directly into an IP network and can be used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.
The Cisco IP phone model terminals can attach to the existing data network infrastructure, via 10BASE-T/100BASE-T interfaces on an Ethernet switch. When used with a voice-capable Ethernet switch (one that understands type of service [ToS] bits and can prioritize VoIP traffic), the phones eliminate the need for a traditional proprietary telephone set and key system and PBX.
Configuring a Cisco ATA 186/188 Device
For further details, refer to the Cisco ATA 186/188 Adaptor Administration Guide.
Step 1
Configure a DHCP server to set up the network configuration for the adaptor.
Note
If your Cisco IP phone network contains a DHCP server, the Cisco ATA adaptor automatically learns its IP address, subnet mask, and network gateway from the DHCP server when the adaptor starts up.
If the DHCP Server is not available, manually assign each network parameters.
Step 2
Configure the TFTP server which will store the configuration files and firmware image.
Note
Use the steps from the "Configuring SIP Parameters via a TFTP Server" section of the Cisco IP Phone 7905 documentation.
Step 3
Download the required files for SIP phone to the root directory of TFTP server. The files required are:
•
Cisco IP Phone 7905 SIP image LD0xxxSIPxxxxxxx.zup .ld1234abcd3456
•
SEP<MACADDR>.cnf.xml e.g (SEP0008a3d31e4a.cnf.xml .. specific for a phone)
or
•
XMLDefault.cnf.xml Default config file downloaded to all adaptors that provide the image.
Step 4
Set up the adaptor configuration, using Configuring Cisco ATA 186/188 Adaptor-Specific Required Information. The adaptor-specific configuration files are added to the root directory.
Step 5
Use the Web page to edit the configuration, or modify the configuration side, and press the button after.
Note
To modify the file for the Cisco ATA 186/188 devices, you also can lift the handset and press the ATA function button to get to the Configuration menu. The Configuration menu allows for inputting key sequences to accomplish minimal configuration changes.
Step 6
Set up the TFTP IP address on the phone. If the adaptor has booted and the network parameters (IPaddr, etc.) are configured, set up the TFTP server IP address if it is not set.
a.
Select NetworkConfig ->AlternatetftpServer.
b.
Set it to Yes.
c.
Select TFTP Server and set the IP address of the TFTP server.
Configuring Cisco ATA 186/188 Adaptor-Specific Required Information
Step 1
Create a ld<lowercase macaddr>.txt file.
Step 2
Convert ld<macaddr>.txt to bin using cfgfmt.exe. Make sure the ptag.dat file is in the same directory as cfgfmt.exe. Run a Windows Command Window at the command prompt >.
cfgfmt ld<macaddr>.txt ld<macaddr>.
The following steps elaborate the contents of ld<lowercase macaddr>.txt file.
a.
Set the tftp server_ip, image ID, and image file name in the adaptor specific configuration file using the following command:
upgradecode:3,0x501,0x0400,0x0100,tftp_server_ip,69,image_id,image_file_name
Example 1 Sample tftp server_ip and image file name
upgradecode:3,0x501,0x0400,0x0100,4.5.6.7,69,0x030218A,LD0101SIP030218A.zup
b.
Enter the UI Password GUI interface password.
c.
Enable or disable the DHCP Server.
d.
Enter the proxy server information (add the Cisco BTS 10200 Registrar or Proxy FQDN).
e.
Enter the UID User Phone number.
f.
Enter the Password Login Authentication information.
g.
Enter the UserLoginId To enable login ID.
h.
Enter the SIPRegOnEnable/Disable registration.
i.
Enter the Codec Set up .
j.
Specify the Timezone.
k.
Enter the DNS1IP.
l.
Enter the UseTftpEnable/Disable TFTP server.
Configuring a Cisco IP Phone 7905
For further details, refer to the Cisco IP Phone 7905 Series Administration Guide.
Step 1
Configure a DHCP server to set up the network configuration for phone.
Note
If your Cisco IP phone network contains a DHCP server, the Cisco IP phone automatically learns its IP address, subnet mask, and network gateway from the DHCP server when the phone starts up.
If the DHCP Server is not available, manually assign each network parameters.
Step 2
Configure the TFTP server which will store the configuration files and firmware image.
Note
Use the steps from the "Configuring SIP Parameters via a TFTP Server" section of the Cisco 7905 documentation.
Step 3
Download the required files for SIP phone to the root directory of TFTP server. The files required are:
•
Cisco 7905 SIP image LD0xxxSIPxxxxxxx.zup .ld1234abcd3456
•
SEP<MACADDR>.cnf.xml e.g (SEP0008a3d31e4a.cnf.xml .. specific for a phone)
or
•
XMLDefault.cnf.xml Default config file downloaded to all phones that provides the image.
Step 4
Set up the phone configuration, using Configuring Cisco ATA 186/188 Adaptor-Specific Required Information. The phone specific configuration files are added to the root directory.
Step 5
Use the Web page to edit the configuration, or unlock the phone to edit configuration.
To edit using the phone:
a.
Use **# to unlock.
b.
Select Highlight to edit the parameter.
c.
Make the changes, and press the SAVE softkey.
Step 6
Set up the TFTP IP address on the phone. If the phone has booted and the network parameters (IPaddr, etc.) are configured, set up the TFTP server IP address if it is not set.
a.
Select NetworkConfig ->AlternatetftpServer.
b.
Set it to Yes.
c.
Select TFTP Server and set the IP address of the TFTP server.
Configuring Cisco IP 7905 Phone-Specific Required Information
Step 1
Create a ld<lowercase macaddr>.txt file.
Step 2
Convert ld<macaddr>.txt to bin using cfgfmt.exe. Make sure the ptag.dat file is in the same directory as cfgfmt.exe. Run a Windows Command Window at the command prompt >.
cfgfmt ld<macaddr>.txt ld<macaddr>
The following steps elaborate the contents of ld<lowercase macaddr>.txt file.
a.
Set the tftp server_ip, image ID, and image file name in the phone specific configuration file using the following command:
upgradecode:3,0x501,0x0400,0x0100,tftp_server_ip,69,image_id,image_file_name
Example 2 Sample tftp server_ip and image file name
upgradecode:3,0x501,0x0400,0x0100,6.7.8.9,0x030218A,LD0101SIP030218A.zup
b.
Enter the UI Password GUI interface password.
c.
Enable or disable the DHCP Server.
d.
Enter the proxy server information (add the Cisco BTS 10200 Registrar or Proxy FQDN).
e.
Enter the UID User Phone number.
f.
Enter the Password Login Authentication information.
g.
Enter the UserLoginId To enable login ID.
h.
Enter the SIPRegOnEnable/Disable registration.
i.
Enter the CODEC Set up .
j.
Specify the Timezone.
k.
Enter the DNS1IP.
l.
Enter the UseTftpEnable/Disable TFTP server.
Provisioning the Cisco IP Phone 7960 for Initial Setup
The following steps are for the initial setup of a Cisco 7960 SIP phone. For further details refer to the Cisco IP Phone 7940/7960 Series Administration Guide.
Step 1
Configure a DHCP server to set up the network configuration for phone.
Note
Use the steps from the "Configuring Network Parameters via a DHCP Server" section of the Cisco 7960 documentation.
If the DHCP Server is not available, manually assign each network parameters.
Step 2
Configure the TFTP server, which will store the configuration files and firmware image.
Note
Use the "Configuring SIP Parameters via a TFTP Server" section of the Cisco 7960 documentation.
Step 3
Download the required files for SIP phone to the root directory of TFTP server. When finished downloading, the following files should appear:
•
OS79XX.TXT (contains an image name)
•
the image file, such as P0S3-04-4-00 or P0S3-04-4-00.bin
Note
The second character in the file above is a zero, not the letter O. For more information about the image name and file, refer to the Cisco 7940/7960 phone configuration guide.
•
SIPDefault.cnf (Phone Global Parameters)
•
SIP<MAC>.cnf (for example, SIP003094C25D40.cnf) (SIP<MAC> is the mac-id)
For more information on the files, refer to the Cisco 7960 SIP phone guide.
Step 4
Set up the phone configuration, using "Creating a Cisco 7960 Phone-Specific Configuration" section.
You can add the phone-specific configuration files to a subdirectory (such as sip_phone). Set tftp_dir: /sip_phone in the SIPDefault.cnf file to allow the phone to get the phone-specific configuration file from that subdirectory (such as the sip_phone file).
Step 5
Unlock to edit configuration .
a.
Select settings-> option 9. If Option 9 (unlock config) is present, select it and enter the password cisco.
b.
Select the parameter to edit, then select EDIT. Make the changes and then choose SAVE.
Step 6
Set up the TFTP IP address on the phone.
Step 7
(Optional) If the phone has booted, but the TFTP server IP address is not automatically obtained by the phone, then set it up as follows:
a.
Select NetworkConfig -> Enable AlternatetftpServer.
b.
Set it to Yes.
c.
Select TFTP Server and set the IP address of the TFTP server.
For more information about the TFTP server, refer to the Cisco 7960 SIP phone guide.
Creating a Cisco 7960 Phone-Specific Configuration
The following task allows you to create a File SIP<upper case MacAddr>.cnf for each phone.
You must prepare the SIP<uppercase MacAddress>.cnf configuration file for the phone, then change the following parameter for line1 to set up a single line on the phone. To set up multiple lines on the phone, add the information to multiple lines.
Step 1
Change line1 Extension\User ID
line1_name: "9025551232"; Line 1
For Extension number line1_name: "51232" ; Line 1
Step 2
Enter the line1 display name.
line1_displayname: "SIP8"
Step 3
Enter the line1_authname used for authenticating all requests from the phone.
line1_authname: "cisco" ; Line 1
Step 4
Enter the authentication.
line1_password: "cisco" ; Line 1
Step 5
Add the Proxy Address, which is the IP address of the CA if it's a SIP subscriber; otherwise, add the Proxy IP address.
Step 6
Enter the Proxy Port; add the CA Port if it's a SIP subscriber. Otherwise, add the Proxy Port.
Step 7
Add the XML file, dialplan.xml, that specifies the dialplan desired to /tftpboot/sip_phone.
dial_template: "dialplan"
Connect Cisco IP Phone 7960 to Cisco BTS 10200
For SIP subscribers, AOR must be provisioned. The user portion of the AOR must be the phone number specified in the linex specification in the phone configuration file. The host portion of the AOR must be the proxy address specified in the linex specification in the phone configuration file (and provisioned in the Serving Domain Name table). For more information, see the Address of Record to Subscriber section in the Cisco BTS 10200 Softswitch SIP Protocol User Guide.
Cisco BTS 10200 Softswitch Phone Mapping
Table 1 shows only the correspondence between the fields in the Cisco BTS 10200 CLI provisioning and the SIP phone configuration file (SIP<macaddr>.cnf). The table does not include all of the provisioning details for either the Cisco BTS 10200 or for the phones.
Table 1 Cisco BTS 10200 Softswitch Phone Mapping
Cisco BTS 10200 Provisioning
|
Cisco ATA 186/188
|
SIP 7905/7912 Phone
|
SIP 7940/7960 Phone (SIP<macaddr.cnf>
|
Auth-realm
add AUTH_REALM_ID=ciscolab;
|
NA
|
NA
|
NA
|
Serving Domain Name
add SERVING_DOMAIN_NAME DOMAIN_NAME=sia-SYS21CA146.ipclab.cisco.com; AUTH_REALM_ID=ciscolab; AUTH_REQD=Y; DESCRIPTION=Cisco Internal;
|
Proxy: sia-domainname. com:5060
|
Proxy: sia-domainname. com:5060
|
proxy1_address:
sia-domainname.com
proxy1_port : 5060
|
Subscriber - DN
add subscriber id=sip_sub4; CATEGORY=INDIVIDUAL; NAME=sipsub4; DN1=4167940001; SUB-PROFILE-ID=sub_profile; TERM-TYPE=SIP; AOR_ID= 4167940001@sia-SYS21CA146.ipclab.cisco.com;
|
UID:"4167940001"
|
UID:"4167940001"
|
line1_name:"4167940001"
|
User Auth
add USER_AUTH AUTH_USER=SIP_7940_ONE; AUTH_REALM_ID=ciscolab; PASSWORD=cisco; AOR_ID=4167940001@sia-SYS21CA146.ipclab.cisco.com;
|
LoginID: SIP_7940_ONE (same as 7960)
UseLoginID: 1
(To use login ID for authentication. LoginID is used if authenticate user information is different from UID.)
PWD:cisco
|
LoginID: SIP_7940_ONE (same as 7960)
UseLoginID: 1
(To use login ID for authentication. LoginID is used if authenticate user information is different from UID.)
PWD:cisco
|
line1_authname:"SIP_7940_ONE"
line1_password:"cisco"
|
In opticall.cfg, the EMS DNS name is used as the TSAP address for the HTTP server.
DNS_FOR_EMS_SIDE_A_CRIT_COM=crit-aSYS21EMS.ipclab.cisco.comadd http-feature-server id=mba; TSAP_ADDR_SIDEA=crit-aSYS21EMS.ipclab.cisco.com; TYPE=HTTP
|
NA
|
NA
|
services_url: "http://crit-aSYS21EMS.ipclab.cisco.com:5252"
|
The Pilot number to the Voice Mail server.
For POTS subscribers, the dial-plan for the trunk is sufficient.
Trunk group type Subscriber is required for voice mail support for Centrex subscribers only.
add subscriber id=UM;category=PBX;dn1=469-555-2001;tgn-id=21;sub-profile-id=sp1;term-type=TG;
|
To access Voice Mail using messages button on SIP phone.
VoiceMailNumber: "4695555555"
The Pilot number can also be specified as a Centrex group extension, if the voice mail system is provisioned as a Centrex SIP trunk.
|
To access Voice Mail using messages button on SIP phone.
VoiceMailNumber: "4695555555"
The Pilot number can also be specified as a Centrex group extension, if the voice mail system is provisioned as a Centrex SIP trunk.
|
To access Voice Mail using messages button on SIP Phone.
messages_uri: "4695555555"
The Pilot number can also be specified as a Centrex group extension, if the voice mail system is provisioned as a Centrex SIP trunk.
|
For regional settings, both phones need to support separate Dial templates to use feature activation/deactivation keys.
|
DialPlan:*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
Add the dial_template defined for regional settings as necessary.
|
DialPlan:*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
Add the dial_template defined for regional settings as necessary.
|
dial_template: "star_region_dialplan"
Add the dial_template defined for regional settings as necessary. The dial-plan template star_region_dialplan.xml must be defined.
|
Services Key: Enabling Feature Activation or Deactivation
This feature can only be used with phones that support HTM services using softkey. The Cisco 7960 is an example of such a phone.
GUI Feature Server Provisioning
This section identifies GUI Feature Server (GFS) provisioning. Cisco BTS 10200 supports SIP client/handset text-based user interface (UI) provisioning for a select set of features, a contrast to many supplementary features supported natively by the SIP client/handset itself. Some features require updating; Cisco BTS 10200 supports SIP clients/handsets to update end user feature access status on the switch network database.
Provisioning refers to activating or deactivating a feature, and setting any applicable Directory Numbers (DNs) associated with the feature. If a SIP handset is used, use the phone's LCD panel as a menu display for feature provisioning. If using a SIP software client, provision the features in the UI display region of the client software.
Configuration
Step 1
Use the -start_gfs command-line parameter for POTS feature in platform configuration file to turn on the GUI Feature Server. This is ON by default.
Step 2
If GUI FS is activated, the -gfsDn parameter to POTS should specify the configured domain name for the GFS that allows communication between EMS and the GFS host.
Office Provisioning
Step 1
Add the HTTP server.
add http-feature-server id=mba;TSAP_ADDR_SIDEA=prica30.ipclab.cisco.com:11227;TYPE=HTTP;
Step 2
Add SCTP association profile.
add sctp-assoc-profile id=sctp_prof_http;bundle-timeout=500; max-assoc-retrans=5;
max-path-retrans=5; retrieve-flag=N; max-rto=6000; min-rto=301; sack-timeout=101;
hb-timeout=1000
Step 3
Add SCTP association.
add sctp-assoc id=assoc_http; sctp-assoc-profile-id=sctp_prof_http;remote-port=5253;
remote-tsap-addr1=priems45; platform-id=FSPTC235; DSCP=AF11; ip-tos-precedence=ROUTINE;
local-rcvwin=18000; max-init-retrans=3; max-init-rto=500; ULP=HTTP;
http-feature-server-id=mba;
Step 4
Put the association IN service.
control sctp-assoc id=assoc_http; target-state=INS; mode=FORCED;
Step 5
Verify SCTP association.
status sctp-assoc id=assoc_http;
SIP Subscriber Services
Individual SIP subscriber provisioning is necessary for delivering GFS features to SIP subscribers, but is outside the scope of the GUI Server provisioning. See the Provisioning a SIP Subscriber section for individual GUI feature subscriber provisioning.
MAC to Subscriber
The MAC to Subscriber (MAC2SUB) table links the MAC address of a device to a subscriber ID. The MAC2SUB table is required to use the GUI interface for feature provisioning on a SIP phone. The table is system generated when the token is used in the Subscriber table, or it can be manually added.
Example 3 MAC to Subscriber example
add mac2sub mac-id=SIP0002B9A74E4C; sub-id=sub1;
Where:
MAC-ID= MAC ID (Mac Address) of the IP phone or device.
SUB-ID= Subscriber ID.
When provisioning SIP subscribers, you also can specify the MAC ID.
Setting Up Services
Step 1
Modify the SIP phone-specific .cnf file on the TFTP server by setting the "services_url" equal to the HTTP address of MBA (along with the port number, such as "services_url=http://1.2.3.4:5252").
Step 2
Re-boot the IP phone(s).
Provisioning a SIP Subscriber
The following steps are required to add a SIP subscriber.
Only the CLI commands for new fields or new tables specific to SIP subscribers are provided in this section. The CLI commands for existing tables such a sub_service_profile, dial_plan, etc. required for the subscriber are not described in this section.
Note
You can use a combination of CLI commands in Step 11 to add the subscriber and all that subscriber's related child tables.
Step 1
Add to AUTH_REALM.
id = ciscolab; description =Cisco Internal;
Step 2
Add the SERVING_DOMAIN_NAME.
The domain name or the IP address in the DomainName field is added. If authentication is required on the phones, set AUTH_REQD=`y'.
domain_name=domainname.com; auth_realm_id=ciscolab; auth_reqd=n; description=Cisco
Internal;
Step 3
Add a SIP subscriber.
id=sip_sub1;CATEGORY=INDIVIDUAL; NAME=SipSub1; STATUS=ACTIVE; LANGUAGE=english;
BILLING-DN=469-555-1111; DN1=469-555-1111; RING-TYPE-DN1=1; SUB-PROFILE-ID=sub_profile;
TERM-TYPE=SIP;
To use the CLI command combination, see Step 11.
Step 4
(Optional) Add AOR2SUB entry.
The Domain portion of the Host ID should be provisioned in the Server Domain table.
aor_id=4695551111@domainname.com; sub_id= sip_sub1;
Step 5
Add the USER_AUTH entry.
This is used only if Auth-Reqd in the serving_domain_name is set to "Y".
auth_user=sipsub1 ;auth_realm_id=ciscolab; aor_id=4695551111@domainname.com;
password=cisco_sipsub1;
Step 6
If the device is not capable of registering itself, a static contact may be used.
static_contact_host=3.4.5.6;static_contact_port=5060; aor_id=4695551111@domainname.com;
user_type=phone;
Step 7
Add MAC2SUB.
Required to use the GUI interface for feature provisioning on SIP phone.
mac_id = SIP0008A3D31E4A;sub_id =sip_sub1;
Step 8
Provision CA-CONFIG to provide min, max and default value for register expires. If not provisioned the default values for each parameter will be used.
For details, refer to the CA-CONFIG SIP Adaptor Configuration Parameters section of the Cisco BTS 10200 Softswitch SIP Protocol User Guide.
type=SIA_REG_MIN_EXPIRES_SECS; datatype=INTEGER ; value=1800;
type=SIA_DEFAULT_REG_EXPIRES ; datatype=INTEGER ; value=3600;
type=SIA_REG_MAX_EXPIRES_SECS; datatype=INTEGER ; value=36000;
Step 9
Set the AORID in the Subscriber table.
change subscriber id= sip_sub1;
aor_id=4695551111@domainname.com;
Step 10
Put AOR in Service.
aor_id=4695551111@domainname.com; status=INS;
Step 11
Step 3 (Subscriber), Step 4 (AOR2SUB), Step 6 (STATIC_CONTACT), Step 7 (MAC2SUB), and Step 9 (Set AOR in SUB table) can be combined by a single CLI command.
id=sip_sub1; CATEGORY=INDIVIDUAL; NAME=SipSub1; STATUS=ACTIVE; LANGUAGE=english;
BILLING-DN=469-555-1111; DN1=469-555-1111; RING-TYPE-DN1=1;
SUB-PROFILE-ID=sub_profile;TERM-TYPE=SIP; aor_id=4695551111@domainname.com;
static_contact_host=3.4.5.6;static_contact_port=5060; user_type=phone; mac_id=
SIP0008A3D31E4A;
Provisioning Subscriber Features
This section describes how to provision Subscriber features. Existing features introduced prior to Release 4.4 are hyperlinked to the Cisco BTS 10200 Softswitch Release 4.4 Provisioning Guide, and the differences for provisioning those features when using SIP are listed in the following feature descriptions.
Activation and Deactivation of Anonymous Call Rejection
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. Provisioning the feature is the same as MGCP when provided by Cisco BTS 10200. ACR is also provided by phone.
For information on provisioning ACR, refer to the Anonymous Call Rejection section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Billing
For detailed information on billing management and data, refer to the Cisco BTS 10200 Softswitch Billing Interface Guide.
CALEA Call Content
CALEA is not available for SIP subscribers.
Call Forwarding
For information about the feature and all of its options, refer to Call Forwarding Features section in the Cisco BTS 10200 Softswitch Release 4.4 Provisioning Guide.
The Call Forwarding feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference between the feature for SIP versus MGCP is 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 will always be played an announcement (if announcements are provisioned) or a re-order tone.
Call Forwarding to an E.164 Number or an Extension Number
In Release 4.4, activation is accomplished using single-stage dialing. This applies to all activation and deactivation.
Calling Name and Number Delivery
These features were introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the features for SIP.
For information on provisioning Calling Name Delivery (CNAM), refer to the Calling Name Delivery section in the Cisco BTS 10200 Softswitch Provisioning Guide.
For information on provisioning Calling Number Delivery (CND), refer to the Calling Number Delivery section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Caller ID Delivery Suppression
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference when provisioning the feature for SIP versus MGCP is as follows:
•
Presentation status from the phone, and single stage digit collection.
For information on provisioning Caller ID Delivery Suppression, refer to the Calling Number Delivery Suppression—Delivery (CIDSD) section and the Calling Number Delivery Suppression—Suppression (CIDSS) section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Called Party Termination
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the features for SIP.
Cisco BTS 10200 Supplementary Vertical Service Code Features
For information on provisioning Vertical Service Codes (VSC), refer to the Vertical Service Code Provisioning section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Customer Access Treatment
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the features for SIP.
For information on provisioning Customer Access Treatment (CAT), refer to the section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Customer-Originated Trace
Use the following CLI for Centrex Subscriber provisioning with the Customer-Oriented Trace (COT) feature:
add cdp id=cdp1;DIGIT_STRING=*57;NOD=VSC;FNAME=COT
For information on provisioning Customer-Originated Trace (COT), refer to the Customer-Originated Trace section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Direct Inward Dialing
There are no special instructions to provision Direct Inward Dialing (DID), other than assigning the DID number to that subscriber as DN1 in the subscriber table.
Direct Outward Dialing
The following subsections identify necessary steps for the Custom Dial Plan (CDP) feature to be offered.
Office Provisioning
Step 1
Provision the feature table.
add/change feature FNAME=CDP; TDP1= COLLECTED_INFORMATION; TID1= CUSTOMIZE_DIALING_PLAN;
TTYPE1=R; FEATURE_SERVER_ID=FSPTC325; DESCRIPTION=Custom Dial Plan Feature;
Step 2
Provision the service table.
add service id=2, FNAME1=CDP;
Do Not Disturb
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference when provisioning the feature for SIP versus MGCP is as follows:
•
Provisioning is the same as MGCP; the difference is activation. Do Not Disturb (DND) can be activated or deactivated from Cisco BTS 10200. Alternatively, activation and deactivation may also be provided through a key on the phone.
For information on provisioning DND, refer to the Do Not Disturb section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Emergency Call
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference when provisioning the feature for SIP versus MGCP is as follows:
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Only E911 (without the suspend procedure for 45 minutes) is supported. Basic 911 with the suspend procedure is not supported.
For information on provisioning Emergency Call (E911), refer to the 911 Emergency Call section in the Cisco BTS 10200 Softswitch Provisioning Guide.
E.164 and Centrex Dialing Plan (Extension Dialing)
Provision the subscriber-service-profile:
add subscriber-service-profile sub_id=sub_1;service-id=2;
Note
CDP feature should be assigned to every CENTREX category users.
Incoming and Outgoing Simulated Facility Group
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the feature for SIP.
For information on provisioning Incoming Simulated Facility Group (ISFG), refer to the Incoming Simulated Facility Group section in the Cisco BTS 10200 Softswitch Provisioning Guide.
For information on provisioning Outgoing Simulated Facility Group (OSFG), refer to the Outgoing Simulated Facility Group section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Multiple Directory Numbers
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference when provisioning the feature for SIP versus MGCP is as follows:
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Ringing part supported by Cisco BTS 10200. Cisco BTS 10200 sends a distinctive alerting request for Call-Waiting scenario; some SIP-Phones interpret it and play distinctive call-waiting tone, while others do not.
For information on provisioning Multiple Directory Numbers (MDN), refer to the Multiple Directory Numbers section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Operator Services (0-, 0+, 01+, 00 Calls)
There is no Cisco BTS 10200 Softswitch Subscriber-specific provisioning involved for Operator Services.
Outgoing Call Barring
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the feature for SIP.
For information on provisioning Outgoing Call Barring (OCB), refer to the Outgoing Call Barring section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Remote Activation of Call Forwarding
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the feature for SIP.
For information on provisioning Remote Activation of Call Forwarding (RACF), refer to the Remote Activation of Call Forwarding section in the Cisco BTS 10200 Softswitch Provisioning Guide.
Type of Service
The ToS value for messages sent to SIP subscribers can be set on a system-wide basis—this applies to all subscribers. The policy is selected in the CA-CONFIG table. The Cisco BTS 10200 reads the values from this table when it starts up. Therefore, changes to the ToS policy for SIP subscribers become effective at the next restart of the Cisco BTS 10200. If the ToS entries are not provisioned in CA-CONFIG table, the following defaults apply:
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Precedence = immediate (010)
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Delay = low (1)
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Throughput = normal (0)
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Reliability = normal (0)
These are the recommended values; these values should be changed only after careful consideration, or if there is a specific need.
Provisioning Network-Level ToS
The ToS value for messages sent to SIP subscribers can be set on a system-wide basis—this applies to all subscribers. The policy is selected in the CA-CONFIG table. The Cisco BTS 10200 reads the values from this table when it starts up. Therefore, changes to the ToS policy for SIP subscribers become effective at the next restart of the Cisco BTS 10200. If the ToS entries are not provisioned in CA-CONFIG table, the following defaults apply:
•
Precedence = immediate (010)
•
Delay = low (1)
•
Throughput = normal (0)
•
Reliability = normal (0)
These are the recommended values; these values should be changed only after careful consideration, or if there is a specific need.
Provisioning Type of Service Default Settings for SIP Subscribers
Note
Note that the `SIA-TRUNK-GRP-LEVEL-SIG-TOS' flag in call agent configuration is used to select between using TOS settings for all SIP trunks or TOS settings for specific SIP trunks.
Step 1
Add the SIA-SIG-TOS-LOWDELAY value.
add ca-config type=SIA-SIG-TOS-LOWDELAY; datatype=BOOLEAN; value=Y;
Step 2
Add the SIA-SIG-TOS-PRECEDENCE.
add ca-config type=SIA-SIG-TOS-PRECEDENCE; datatype=INTEGER; value=2;
Step 3
Add the SIA-SIG-TOS-RELIABILITY value.
add ca-config type=SIA-SIG-TOS-RELIABILITY; datatype=BOOLEAN; value=N;
Step 4
Add the SIA-SIG-TOS-THROUGHPUT value.
add ca-config type=SIA-SIG-TOS-THROUGHPUT; datatype=BOOLEAN; value=N;
Phone-Based Features
Phone-based features are provided by the SIP phone, which require provisioning on the phone.
There are some features that the phone provides standalone, without Cisco BTS 10200 support.
The Cisco BTS 10200 Softswitch supports interface requirements (such as Re-INVITE support) that are necessary to operate features from the SIP phones, including but not limited to:
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Call Hold and Resume
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Call Waiting
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Three-Way Calling
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Cancel Call Waiting
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Call Waiting Caller ID
•
CODEC Up-speeding
For information on provisioning these features, refer to the SIP phone documentation.
Jointly-Provided Features
In addition to the Softswitch-based and phone-based features, Release 4.1 also offered jointly-provided features. These are features provided jointly by the phone and by the Cisco BTS 10200. To use these features, you must provision both the phone and the Cisco BTS 10200.
Session Timer
Step 1
Change the softswitch trunk group profile ID.
softsw_tg_profile id=<profile_id>; SESSION_TIMER_ALLOWED=Y;
Step 2
Add the CA-CONFIG session-expires value.
add ca-config type=session-expires;data-type=INT; value=3600;
Step 3
Add the CA-CONFIG min-se value.
add ca-config type=min-se;data-type=INT;value=900;
Call Transfer (Blind and Attended) via Refer Feature
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. The difference when provisioning the feature for SIP versus MGCP is as follows:
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Call transfer on both the Cisco IP Phone 7905/7912 and the Cisco IP Phone 7940/7960 is done using soft keys. 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. For information on provisioning the Refer feature, see the Refer section of the Cisco BTS 10200 Softswitch Provisioning Guide.
Distinctive Ringing
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the feature for SIP.
Distinctive Ringing for Centrex DID Calls
This feature was introduced in a previous Cisco BTS 10200 Softswitch release. There are no differences when provisioning the feature for SIP.