Use a Cisco Unified SRST router, rather than Unified CME in SRST mode, for the following deployment scenarios:
•For supporting a maximum of 1,500 phones on a single SRST router.
•For up to 3,000 phones, use two SRST routers. Dial plans must be properly configured to route the calls back and forth between the SRST routers.
•For simple, one-time configuration of basic SRST functions.
•For SRTP media encryption, which is available only in Cisco Unified SRST (Secure SRST).
•For support of the Cisco VG248 Voice Gateway.
For routing calls to and from phones that are unreachable or not registered to the SRST router, use the alias command.
Enhanced Survivable Remote Site Telephony
Enhanced Survivable Remote Site Telephony (E-SRST) simplifies the deployment of Cisco Unified CME running SRST in the branch. E-SRST architecture is built on top of Survivable Remote Site Voicemail (SRSV). During normal operation, E-SRST utilizes Cisco Unified Messaging Gateway E-SRST located at headquarters site to regularly retrieve configurations (for example, calling search space, partition, hunt group, call park, call pickup, and so forth, if configured) from Cisco Unified CM and upload them to provision the branch router with similar functionality for use in SRST mode. Thus, E-SRST reduces manual configuration required in Unified CME running SRST and enables users to have a similar calling experience in both SRST and normal modes.
What’s new with SRST 9.x
· Support for Cisco Unified IP Phone 6901, 6911, 6921,6941, 6945 & 6961 SIP IP Phones
· Forced Authorization Code (FAC)
· Normalized +E.164 support for SRST
· Improved deployment flexibility with support for SSL VPN client on Cisco Unified IP Phones (SCCP)
· Customizable Programmable Line Keys, Button Layout Control
· ISDN overlap sending on PRI/BRI
E-SRST Overview
· The Survivable Provisioning Infrastructure
o Automatically provisions remote branch sites
o Always in-sync with the Centralized Communications Manager and pushes the updates to the Branch routers
o Automatically synchronizes moves, adds, and changes across from CUCM to Branch ISR
· Phone User Experience
o Device display and phone user operations are identical in survivable mode
· Call Treatment:
o Access and security control
o Digit manipulation for both incoming and outgoing calls
· Management:
o GUI interface for provisioning, monitoring, report and troubleshooting.
o On Demand info sync with CUCM
https://supportforums.cisco.com/videos/3031
http://www.cisco.com/go/srst
https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
Sample Configuration
SRST Sample Configuration .txt
Cisco Secure SRST Configuration Example
Unified SRST Interactive Voice Network Configuration Example
SRST Voice Mail Interactive Voice Network Configuration Example
Common Show & Debug Commands
Show Commands
Command | Purpose |
show run | Displays the configuration. |
show call-manager-fallback all | Displays the detailed configuration of all the Cisco IP Phones, voice ports, and dial peers. |
show call-manager-fallback dial-peer | Displays the output of the dial peers of the SRST Catalyst 4224. |
show call-manager-fallback ephone-dn | Displays the Cisco IP Phone destination number. |
show call-manager-fallback voice-port | Displays output for the voice ports. |
show ephone | Displays Cisco IP Phone output. |
show ephone-dn | Displays the Cisco IP Phone destination number. |
show ephone summary | Displays a summary of the Cisco IP Phone output. |
show voice port summary | Displays a summary of all voice ports. |
show dial-peer voice summary | Displays a summary of all voice dial peers. |
Debug ephone register
Debug ephone state
Debug ephone details
Management (SNMP)
Cisco Unified SRST SNMP MIB Release 4.0 Guide
Cisco SRST SNMP MIB Release 3.4 Guide
For general SRST product related questions, please contact the Cisco Product team.
mailto:access-ccme-cue
Model | Mbps | CME Max User | SRST Max User |
1841 | 2 | 0 | 0 |
1941 | 25 | 35 | 35 |
2801 | 2 | 25 | 24 |
2901 | 25 | 35 | 35 |
2811 | 4 | 35 | 36 |
2911 | 35 | 50 | 50 |
2821 | 8 | 50 | 48 |
2921 | 50 | 100 | 100 |
2851 | 12 | 100 | 96 |
2951 | 75 | 150 | 250 |
3825 | 22 | 175 | 336 |
3925 | 100 | 200 | 730 |
3845 | 44 | 250 | 720 |
3945 | 150 | 300 | 1200 |
SRST vs CME in SRST
SRST vs E-SRST
E-SRST combines all the benefits of SRST, but also includes the following:
• Enhanced user experience in failover mode by maintaining phone displays and providing full call-control features
• Easy to use GUI interface to provision, monitor, report, and troubleshoot remote sites
• Automatic synchronization with Cisco Unified Communications Manager for additions, deletions, and modifications of users and phones
• Calling-rule restrictions continued in failover mode
CME/SRST Release Version | IOS Version |
CME/SRST 4.1 | 12.4(15)T |
CME/SRST 4.2 | 12.4(11)XW3 |
CME/SRST 7.0 | 12.4(20)T |
CME/SRST 7.1 | 15.0(1)M2 (Golden Release, recommended for most customers |
CME/SRST 8.0 | 15.1.1T |
CME/SRST 8.1 | 15.1.2T |
CME/SRST 8.5 | 15.1.3T |
Features by Cisco Unified SRST Software Version | ||
Cisco Unified SRST | Cisco IOS Release | Enhancements or Modifications |
Version 9.0 | 15.2(2)T | •Support for Cisco Unified 6901 and 6911 SIP IP Phones•Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones•Support for Cisco Unified 8941 and 8945 SIP IP Phones •Multiple Calls Per Line •Voice and Fax Support on Cisco ATA-187 |
Version 8.8 | 15.2(1)T | Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones |
Version 8.0 | 15.1(1)T | Beginning with Cisco IP Phone firmware 8.5(3) and Cisco
IOS Release 15.1(1)T, Cisco SRST supports SIP signaling over UDP, TCP,
and TLS connections, providing both RTP and SRTP media connections based
on the security settings of the IP phone. For more information, see the
following sections:•Signaling Security on Unify SRST – TLS •Media Security on Unify SRST – SRTP •Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST |
Version 7.0/4.3 | •Configuring Eight Calls per Button (Octo-Line)•Configuring Consultative Transfer | |
Version 4.2(1) | Enhanced 911 ServicesThe following new features are included: •Assigning ERLs to zones to enable routing to the PSAP that is closest to the caller. •Customizing E911 by defining a default ELIN, identifying a designated number if the 911 caller cannot be reached on callback, specifying the expiry time for data in the Last Caller table, and enabling syslog messages that announce all emergency calls. •Expanding the E911 location information to include name and address. •Adding new permanent call detail records. |
Nice post!!!
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