Q. What is Cisco
® Unified Survivable Remote Site Telephony (SRST) and Cisco Unified Enhanced SRST
A. In a centralized Cisco Unified Communications Manager or Business Edition environments, when IP phones lose connectivity to Cisco Unified Communications Manager or Business Edition because the WAN is down or the application is unreachable, IP phones in remote branch offices or teleworker homes lose call-processing capabilities. SRST provides basic IP telephony backup services so that IP phones can fall back to the local router at the remote site when connectivity is lost. SRST takes advantage of existing Cisco IOS
® Software features to provide basic telephony services such as off-net calls to 911, calls within a remote site, or calls between sites through the public switched telephone network (PSTN). The application is ideal for enterprises for cost-effective deployment of IP telephony in their remote locations.
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 or Business Edition for additions, deletions, and modifications of users and phones
● Calling-rule restrictions continued in failover mode
Table 1. Additional Features Differences Between Cisco Unified SRST and Cisco Unified Enhanced SRST
(In Addition to Those Listed Above)
Additional Feature Differences
Cisco Unified E-SRST
Cisco Unified SRST
• 450 Total phones are supported
• 1500 Total phones are supported
• Cisco VG248 48-port analog phone gateway is not supported for fallback in E-SRST
• Cisco VG248 48-port analog phone gateway is supported for fallback in SRST mode
• Alias command is not supported
• Alias command is supported
• Enhanced features such as call park, call pick, ephone-template, video survivability, BLF, shared-line are supported with E-SRST
• Only basic features are supported with SRST. Please refer to SRST Admin guide for a list of supported SRST features (see below)
• Class of restriction is a supported feature in E-SRST
• Class of restriction is not supported in SRST
• SRST Manager supports provisioning of E-SRST
• SRST Manager supports provisioning of SRST from SRST Manager 9.0
• Supports Cisco Hosted Collaboration Solution (HCS)
Q. Is special hardware required for SRST or E-SRST?
Q. What features does E-SRST support that SRST does not?
A. E-SRST enabled through Cisco Unified SRST Manager supports the following features:
● Automatic provisioning: Integrated with Cisco Unified Communications Manager, Cisco Unified E-SRST offers automatic provisioning of branch-office routers with the aid of the Cisco Unified SRST Manager. Cisco Unified E-SRST supports centralized Cisco Unified Communications Manager deployments and Cisco Unified Business Edition deployments.
● Automatic synchronization with the central site: The Cisco Unified SRST Manager synchronizes the configuration from the central site to the branch office on a schedule without manual intervention. The synchronization also can be triggered manually on a temporary basis. Branch-office site adds, moves, and changes are handled centrally on the Cisco Unified SRST Manager.
● Consistent device layout: The consistent design of the phones results in better user experiences during service outages. Phone displays and basic functions including extensions, soft-key templates, phone types, etc. are carried over in survivable mode.
● Full call-control features during failover: The Cisco Unified SRST Manager synchronizes the system behavior for the following features (in addition to the features listed for Cisco Unified SRST):
- Call-forward no-answer, call-forward all, and call-forward busy
- Time-of-day routing
- Calling route restrictions for both incoming and outgoing directions
- Hunt groups
- Call park and call pickup
Q. What routing platforms support Cisco Unified SRST and E-SRST?
A. Cisco Unified SRST and Unified E-SRST are available on the Cisco 880, 2900, and 3900 Series and the Cisco 4451 Integrated Services Routers.
Q. Where can I find information about router platforms, the maximum number of phones or virtual ports, and recommended memory requirements for SRST?
Q. Where can I get the SRST files and SRST and Cisco IOS Software versions and supported features?
The SRST function is part of the Cisco IOS Software itself. Multiple Cisco IOS Software packages that support SRST are available. Visit the Cisco IOS Software feature navigator to choose the right Cisco IOS Software image for your router and the functions you need:
Cisco Unified SRST routers do not need phone software files because all the phones load their software directly from the Cisco Unified Communications Manager or Cisco Unified Business Edition. You need extra files on your SRST router only if you want to use the music-on-hold (MOH) file that Cisco provides and the H.450 call-transfer application. These files are available at:
(look for srst-3.3 zip).
Q. Which Cisco IOS Technology Package do I need to run SRST and E-SRST on the Cisco Integrated Services Routers Generation 2 (ISR G2) routers?
A. The Unified Communications (UC) Technology Package is required to enable SRST and E-SRST on the ISR G2 routers. All ISR G2 routers (Cisco 2900 and 3900 Series ISRs) are shipped with a universal IOS image. A UC technology license pack needs to be installed on the router to provision SRST and E-SRST. The Cisco 880 Series Integrated Services Router supports only a single feature set with Advanced IP services software.
Q. What are the recent feature additions to Cisco SRST and E-SRST?
A. The recent feature additions are listed in the chart below.
Recent Feature Additions to Cisco SRST and E-SRST
Cisco SRST and E-SRST Version Number
• Cisco Unified SRST is now supported on ISR 4451 with XE 3.9
• Video survivability for the Cisco 8900, 9900 and Cisco Jabber endpoints(E-SRST)
• Shared line support (E-SRST)
• Busy lamp field support (E-SRST)
9.1 and 9.5
• Trunk to trunk transfer block for toll fraud prevention on SIP phones
• Forced Authorization Code (FAC)
• Improved deployment flexibility with support for SSL VPN client on Cisco Unified IP Phones (Skinny Client Control Protocol [SCCP])
• Support for normalized +E.164 dialing is also introduced
• Enable the configuration for the differentiated services code point (DSCP) values for media packets and signaling packets on the system level
• Introduces the voice translation rules and profiles allowing the administrator to manipulate the calling numbers, called numbers, or redirected called numbers for a voice call.
• Uses the translation rule to implement the +E.164 format
Q. Do I need to buy a license in order to use Cisco Unified SRST and Unified E-SRST?
A. Yes, you need to buy a separate feature license to run SRST in a production network. You can either buy the SRST feature license individually and a supported router to run the SRST feature or buy the integrated services router SRST bundles that come with the SRST feature license included.
Q. Is there any way to save money when ordering routers and the SRST license together?
Q. If I am upgrading my Cisco Integrated Services Router (ISR) from Generation 1 (G1) to Generation 2 (G2), can I reuse my existing SRST licenses?
A. No. New licenses are needed.
Q. What version of Cisco Unified Communications Manager and Business Edition does Cisco Unified SRST and Cisco Unified E-SRST support?
A. All versions of Cisco Unified SRST work with all versions of Cisco Unified Communications Manager. It is not the Cisco Unified Communications Manager and Business Editions version that affects the support, but the phone load or firmware version (associated with the Cisco Unified Communications Manager/Business Edition release).
Cisco Unified E-SRST supports the same versions of Cisco Unified Communications Manager/Business Edition as SRST as well.
Q. Which IP phones or phone loads are supported and tested with Cisco Unified SRST and Unified E-SRST?
Q. Is Jabber Supported?
A. Jabber for Windows and MAC is supported with SRST and E-SRST 10.0.
Octo-Line and Consultative Transfer
Q. Can you explain the octo-line feature?
A. This feature supports up to eight active calls, both incoming and outgoing, on a single button. Eight incoming calls to an octo-line directory number ring simultaneously. After an incoming call is answered, the ringing stops and the remaining seven incoming calls hear a call-waiting tone.
After an incoming call on an octo-line directory number is answered, the answering phone is in the connected state. Other phones that share the directory number are in the remote Multiline state. A subsequent incoming call sends the call-waiting tone to the phone connected to the call, and sends the ringing tone to the other phones that are in the remote Multiline state. All phones sharing the directory number can pick up any of the incoming unanswered calls.
When multiple incoming calls ring on an octo-line directory number that is shared among multiple phones, the ringing tone stops on the phone that answers the call, and the call-waiting tone is heard for other unanswered calls. The multiple instances of the ringing calls are displayed on other ephones sharing the directory number. After a connected call on an octo-line directory number is put on hold, any phone that shares this directory number can pick up the held call. If a phone is in the process of transferring a call or creating a conference, other phones that share the octo-line directory number cannot “steal” the call.
Q. Can you explain consultative transfer?
A. Before Cisco Unified SRST 7.0, the consultative transfer feature played a dial tone and collected dialed digits until the digits matched the pattern for consultative transfer, blind transfer, or PSTN transfer blocking. The after-hours blocking criteria were applied after the consultative transfer digit collection and pattern matching.
The new feature modifies the transfer digit-collection process to make it consistent with Cisco Unified Communications Manager. This feature is supported only if the
transfer-system full-consult command (default) is specified in call-manager-fallback configuration mode and an idle line or channel is available for seizing, digit collection, and dialing.
The new enhancement, by default, collects the transfer digits from the new call leg. You also can configure the system to collect the transfer digits from the original call leg.
Q. Can you explain the E-911 feature support in Cisco Unified SRST and E-SRST?
A. With Enhanced 911, the location of a caller is identified by the emergency workers at the Public Safety Answering Point (PSAP) using a database called the Automatic Location Identifier (ALI) database. Businesses generally have a few special numbers called Emergency Line Identification Numbers (ELINs) registered with the ALI database. The total number of registered numbers is determined by local laws, depending upon the geographic area of the business.
With the E-911 for SRST and E-SRST feature, the calling number sent in outgoing emergency calls is replaced by an ELIN so that emergency workers can identify the caller’s location. The E-911 feature also allows emergency operators to call back to connect to the last caller who called the emergency services.
Q. How much does E-911 for SRST and E-SRST support cost?
A. There is no charge for E-911 for SRST and E-SRST support on the Cisco Unified router, but there could be charges for the PSTN trunks to connect to the E-911 network.
To deploy E-911 for SRST and E-SRST support, you need to buy only the right Cisco IOS Software image with UC technology license package and SRST feature licenses for the number of phones you have along with the right modules for the PSTN trunk for E-911 connectivity.
Q. What PSTN trunks do I need for E-911 for SRST and E-SRST support?
A. E-911 for SRST and E-SRST s supported for Centralized Automated Message Accounting (CAMA) and ISDN Primary Rate Interface (PRI) trunks.
Q. With the support for the E-911 feature in SRST and E-SRST, do I still need Cisco Emergency Responder?
A. Yes. E-911 for SRST and E-SRST works only in the fallback mode. During normal operation, when the WAN link is up, you need to use Cisco Emergency Responder, which works with Cisco Unified Communications Manager and Business Edition.
Q. With the E-911 for SRST and E-SRST feature, can I support all the Cisco Emergency Responder features in the fallback mode?
A. Both Cisco Emergency Responder and E-911 for SRST and E-SRST support location tracking of a phone based on the IP address subnet that is assigned to each phone.
The Cisco Emergency Responder using the Cisco Discovery Protocol also provides automatic location tracking for phones based on the switch port that the phone is plugged into. The E-911 for SRST feature does not support per-phone ERL assignment to an individual SCCP or SIP phone.
Q. What types of phones and fallback modes are supported with the E-911 for SRST and E-SRST feature?
A. E-911 is supported for SCCP, SIP, and analog phones. It is supported in the SIP, Media Gateway Control Protocol (MGCP), and H.323 fallback modes.
SRST and E-SRST Fallback Support
Q. What if the WAN link fails when an IP phone in the branch office is calling another IP phone or analog phone in the same branch office? Because the IP phone is under the supervision of the central-site Cisco Unified Communications Manager or Business Edition, what happens to the call?
A. When the WAN link fails, the existing calls between IP phones are maintained until either side hangs up. Any call legs that involve H.323 (that is, calls using analog phones not configured for SCCP control and H.323 gateways; for example, the Cisco VG224 Analog Phone Gateway) may get dropped within 3 minutes of the WAN failure unless the
no h225 timeout keepalive command is configured. Also, the IP phones that are in an active call at the time of the fallback are not re-homed to the SRST/E-SRST router for the duration of the call, and thus cannot receive new calls from other phones within the same branch office until they do register with the router.
Q. Active calls on SRST/E-SRST-enabled H.323 gateways and analog phones are preserved for about 3 minutes before they are dropped. Why 3 minutes?
A. H.245 TCP keepalives associated with the H.323 call leg are sent once every minute. The calls are dropped after the H.245 TCP keepalive timer expires. When TCP keepalives are sent out without receiving four acknowledgments in sequence (a process that takes up to 5 minutes), the call leg including the Real-Time Transport Protocol (RTP) voice path is dropped. The time actually varies and it also depends on phone type. The following observations were noted in the lab when calls were made from one branch office to another and then the Cisco Unified Communications Manager link was shut down for longer than 10 minutes:
● Active calls from IP phone to IP phone last for the duration of the conversation
● Active calls from IP phone to the PSTN last about 3 minutes before the call is cut off
● This duration addresses a vast majority of typical voice conversations
● Calls are dropped after four H.225 keepalives expire on the gateway (timer is configurable with Cisco CallManager 4.01 and later and all versions of Cisco Unified Communications Manager)
● Cisco Unified Communications Manager resets active calls when the Cisco Unified Communications Manager WAN is restored
● IP phones cannot start placing new calls until they are registered to the Cisco Unified SRST router
Q. Is there any way to keep the calls for H.323 gateways or analog phones for longer than 3 minutes after the WAN link fails?
A. Yes, if you use the H.323 VoIP Call Preservation for WAN Link Failures feature, the router does not automatically terminate calls. With this feature you can preserve existing H.323 calls at the remote site if an outage occurs by disabling the H.225 keepalive timer with the
no h225 timeout keepalive command.
Q. Are other enhancements available with H.323 VoIP Call Preservation for WAN Link Failures?
A. Cisco Unified SRST 4also addresses some of the rare scenarios in which the call preservation did not work. Support was also added for dealing with WAN instability when the WAN link breaks and restores multiple times within a short period of time.
Observations During WAN Outage
Q. How can I tell if phones are in SRST/E-SRST mode?
A. When phones are in SRST mode, depending on the configuration, you will see either “CM Fallback Service Operating” or a custom message on the phone display.
Q. How long does it take for IP phones to fall back to the Cisco Unified SRST/E-SRST router?
A. You get an “instant” failover if and only if:
● Failover occurs when the phones have a hot standby TCP socket already open to the failover device (SRST or the second Cisco Unified Communications Manager, Business Edition or Cisco HCS).
● Failover occurs when the TCP connection is explicitly closed by a TCP FIN, RST, or Internet Control Message Protocol (ICMP) host unreachable, so that the phone is not replying until it times out: The phone sends station keepalive messages to its primary and backup Cisco Unified Communications Manager servers. The phone knows the TCP connection of its backup server is up because of the keepalive messages, and attempts registration when needed. The following fields are recommended default values (though configurable):
- StationKeepAliveInterval: 30 seconds
- Station2ndKeepAliveInterval: 180 seconds
Following are the three scenarios of failovers:
1. Failover occurs when the Cisco Unified Communications Manager/Business Edition server to which the phone is registered stops working: If the active Cisco Unified Communications Manager is manually stopped, the failover is immediate because the Cisco Unified Communications Manager closes its TCP connections, causing the phone to register with its designated standby (or backup) server immediately.
2. Failover occurs when Cisco Unified Communications Manager/Business Edition process locks up: This scenario is the worst scenario; if it occurs, failover could potentially take 90 seconds because the phone makes three station keepalive attempts spaced 30 seconds apart as default (30 seconds is configurable on the Cisco Unified Communications Manager, however).
3. Failover occurs when a TCP failure occurs: TCP failure could be the result of a router or switch going down, or the server itself going into complete failure mode. As soon as the phone sends its first station keepalive message after the TCP connection is down, it sends TCP retries for approximately 20 to 25 seconds. After that, the phone attempts registration with its designated standby (or backup) server.
IP Phone in a Remote Site to IP Phone at Headquarters
Q. If an IP phone in a remote site is in a session with an extension at headquarters and the WAN link fails, what will happen?
A. If the call media is going over the WAN link that failed, then the call is terminated. If the call media is going over the PSTN, the call can be maintained until either party hangs up the phone, using the No Timeout for Call.
IP Phone in One Remote Site to IP Phone in Another Remote Site
Q. When the WAN link is down, will the IP phones at other locations or other remote sites within the same Cisco Unified Communications Manager or Business Edition still be able to make calls to the site with the failed WAN link?
A. When the WAN link is down, IP phones at other remote sites cannot know if the failed site is in SRST/E-SRST mode, and will receive fast-busy signals when trying to call phones at the SRST/E-SRST site. You have to use the PSTN to reach phones at the SRST site. You can configure Cisco Unified Communications Manager and Business Edition to route calls through the PSTN when the WAN link is down, but Cisco Unified Communications Manager/Business Edition cannot differentiate whether the failure at the remote site was caused by loss of WAN connectivity or a failure on the IP phone.
IP Phone at Headquarters to IP Phone in a Remote Site
Q. What will happen when the IP phones at a central site try to call a remote site when the WAN link to that office is down?
A. When the WAN link fails, the Cisco Unified Communications Manager detects that the IP phones at the remote site are unreachable based on loss of the keepalive signals from the IP phones. Users attempting to call these phones receive a busy or fast-busy signal by default.
You can use the Call Forward Unregistered feature to forward calls to the remote site using the PSTN as an alternate route. With this feature, calls to unregistered destinations can be forwarded to an alternate destination. During WAN failure cases, the remote-site phones become unregistered and the Call Forward Unregistered feature converts the remote-site extensions dialed at the headquarters to a full E.164 number and routes these calls over the PSTN.
Analog Phone in the Branch Office to IP Phone through the PSTN
Q. When a call is placed within the same branch office during a WAN link failure (the call is in progress and the WAN link recovers), will the central-site Cisco Unified Communications Manager try to take control of the existing call?
A. The existing call is maintained by the SRST/E-SRST-enabled router unless the user terminates it; after that the IP phone “re-homes” back to the Cisco Unified Communications Manager or Business Edition or Cisco HCS.
Q. Several branch offices are in a metropolitan (metro) area connected through gigabit LAN connections and a centralized Cisco Unified Communications Manager model is used; the SRST/E-SRST -enabled routers have no WAN connections, and the only connections of the routers in branch offices are Ethernet and foreign-exchange-office (FXO) interfaces for 911 and off-net dialing. If a gigabit LAN link fails, will the SRST/E-SRST feature work?
A. Yes, it will. The key in having the phones switched to register with an SRST/E-SRST router is a loss of keepalive packets from the Cisco Unified Communications Manager/Business Edition. When the phones miss three keepalive packets, they register with the local router - the SRST/E-SRST router - which is the default router.
Q. Do I need to configure a MAC address for SRST/E-SRST mode?
A. You do not need to configure a MAC address if the IP phones are already configured by Cisco Unified Communications Manager for SRST Mode. SRST Manager takes care of the MAC address and phone provisioning configuration in E-SRST mode. When the connection to the Cisco Unified Communications Manager is down, or after the IP phones miss three keepalive packet exchanges with the Cisco Unified Communications Manager, IP phones register with the SRST/E-SRST router and also provide the SRST router with all related configuration that is contained in their memory.
Q. Are both the Cisco Unified Communications Manager Express and SRST/E-SRST functions included in the code, even if I am using just SRST/E-SRST? Does it cost the same for license fees for both functions?
A. Although both functions are in the code base, you can activate only one of the functions at any given time. If you turn on SRST/E-SRST, you will not be able to activate Cisco Unified Communications Manager Express.
Q. If I can conduct WAN calls, what happens when the WAN goes down?
A. In the SRST/E-SRST mode, the phones register with the local router, which provides the call processing. Calls from the site then use the PSTN for access. The phone display indicates that you are in a “Cisco Unified Communications Manager Fallback Service Operating” mode, and you may have to dial the full phone number instead of just an extension number.
Q. What video features does Cisco Unified SRST/E-SRST support?
A. Cisco Unified E-SRST supports video survivability on SIP phones.
Q. What phones can I use for video calls with Cisco Unified SRST/E-SRST?
A. Cisco Unified SRST does not support video survivability but Cisco E-SRST does for select endpoints such as the Cisco 8900 and 9900 IP phones and Cisco Jabber. Cisco Jabber requires connectivity to the centralized IM and presence server today.
Q. What is fax Passthrough using SCCP and analog-telephone-adapter support?
A. Cisco Unified SRST supports Fax Passthrough using Cisco VG224 Voice Gateways and analog telephone adaptors using the SCCP protocol. With this feature G.711 media is used between the Cisco Unified Communications Manager Express and the Cisco VG224 or analog telephone adaptor and the fax tones are passed in band between the Cisco Unified Communications Manager Express router and the Cisco VG224 or analog telephone adaptor.
Q. Explain the Secure SRST feature support in Cisco IOS Software routers?
A. Cisco routers can function in Secure SRST mode, which activates when the WAN link or Cisco Unified Communications Manager/Business Edition or Cisco HCS is not available. The Secure SRST feature is a software feature added to Cisco IOS Software advipservicesk9 and adventerprisek9 images to provide secure calls for IP phones in SRST mode with authentication and encryption support for both signaling and media transmission. Signaling encryption is done with transparent LAN services (TLS) for call setup information, media encryption keys, dual-tone-multifrequency (DTMF) tones, and personal identification numbers (PINs), etc. Media encryption uses Secure Real-Time Transport Protocol (SRTP) to protect voice conversations.
Secure phones connect to port x+443 (default 2443), where x is the TCP port set in the IP
source-address command under
call-manager-fallback configuration mode.
You can tell if the call is secure by checking to see if a secure lock icon shows on the IP phone display. When the WAN link or Cisco call control is restored, Cisco Unified Communications Manager, Business Edition or Cisco HCS resumes secure call-handling capabilities.
Q. What security features does E-SRST support that SRST does not?
A. Secure SRST is not supported in E-SRST mode today.
Q. Which Cisco IP Phones does Secure SRST support?
Q. What are the supported Secure SRST features?
A. Following are the supported Secure SRST features:
● Basic call
● Call transfer (consult and blind)
● Call forward (busy, no answer, and all)
● Shared line (IP phones)
● Hold and resume
● Hold and pickup
Note: Calls are secure only between IP phones on an SRST router.
Q. Which features are not supported by Secure SRST?
A. The following Secure SRST voice security features are not supported:
● Calls across H.323, MGCP, and SIP endpoints
● Secure transcoding or conferencing (three-way calling)
● Secure SRST interworking with Cisco Unity Express
● Hot Standby Router Protocol (HSRP)
● Advanced-integration-module (AIM) VPN hardware acceleration
Q. Is there any performance effect on a Cisco Unified SRST router running the Secure SRST feature?
A. SRTP Media Encryption has minimal effect on the number of calls unless it is deployed with the G.711 codec. However, TLS is CPU- and memory-intensive for supporting SRST phone registration, and the number of concurrent secure calls supported is less than the number for nonsecure calls.
Q. What is the music-on-hold (MOH) live-feed support?
moh-live command provides live-feed MOH streams from an audio device connected to an ear & mouth (E&M) or FXO port to Cisco IP Phones in SRST/E-SRST mode. If you use an FXO port for a live feed, the port must be supplied with an external third-party adapter to provide a battery feed. Music from a live feed is obtained from a fixed source and is continuously fed into the MOH playout buffer instead of being read from a flash memory file. Live-feed MOH can also be multicast to Cisco IP Phones.
Q. What kind of external third-party adapter can I use for MOH live feed off an FXO port?
Cisco recommends part number ST-TC1 from RDL because it has been tested in the lab. Refer to the following URL for more information:
Search the web for “ST-TC1 telephone system coupler” for a local supplier.
SRST and SIP
Q. What is a SIP redirect server in Cisco IOS Software?
A. It is a SIP registrar that enables SIP registrar functions on the gateway to accept incoming SIP REGISTER messages and then auto create voice-over-IP (VoIP) dial peers; it is also an IP-to-IP redirect server for local SIP-to-SIP phone calls.
Q. What is the SIP Gateway Enhancement feature?
A. When SIP is used for on-net calls to integrate with SIP networks, the SIP Gateway Enhancement feature provides the following:
● Symmetric gateway-to-gateway out-of-band (OOB) DTMF relay
● Support for unsolicited NOTIFY OOB DTMF relay
● SIP Registrar: No security or authentication support
● Unsolicited NOTIFY message-waiting indicator (MWI)
● Lighting of the MWI upon receiving NOTIFY message
● Support for 300 multiple choices to SIP security parameter index (SPI)
Note: SCCP IP phones cannot do in-band digit relay or support RFC 2833, and the Cisco Unity system working in SIP mode cannot do the full SIP SUBSCRIBE/NOTIFY for MWI.
Q. Can I use multiple SRST routers to support more phones?
A. Yes, you can run multiple routers as SRST routers to support more phones than the supported limit on each router. And also note that you need to carefully plan and configure dial peers and dial plans for call transfer and call forwarding to work properly.
Q. What does the voicemail command do in SRST?
A. The voicemail command in call-manager-fallback mode is used to configure the telephone number that is speed-dialed when the message button on a Cisco IP Phone is pressed. The same voicemail telephone number is configured for all Cisco IP Phones connected to the router. For example:
The number 914085252222 is called when the Cisco IP Phone Messages button is pressed to retrieve messages.
Q. What is the support for voicemail integration with the Cisco Unity server through analog or DTMF?
A. SRST uses the same in-band analog or DTMF voicemail integration method that Cisco Unified Communications Manager Express uses to allow call-forward busy, call-forward no-answer, or call-forward all to the Cisco Unity server through analog or DTMF through the PSTN. An incoming call can be forwarded to the Cisco Unity voicemail server when call-forward busy, call-forward no-answer, or call-forward all is configured in the SRST router. However, MWI integration is not yet supported in SRST. You can rely on the “Missed calls” shown on the phone display to check for your voicemail. Note that FXO hairpin forwarded calls to voicemail must have disconnect supervision from the central office.
Q. Is there a way to maintain call detail records (CDRs) during SRST/E-SRST mode and synchronize records with Cisco Unified Communications Manager records?
A. The SRST/E-SRST router provides CDRs in two formats - syslog and RADIUS. In both formats, you need an external server to collect the CDR records from the SRST/E-SRST router.
Cisco Unified Communications Manager and Business Edition are not designed to process the CDR records generated by the router during fallback. Calls that are placed through the PSTN receive billing information only by the PSTN billing system. You can use CiscoWorks Voice Manager to export the call-history log to a file that can then be processed in CDR format. CDR records can also be logged onto a syslog or a RADIUS server.
Q. Can I run the MGCP Gateway Fallback feature with SRST/E-SRST on the same router?
A. Yes, MGCP and SRST/E-SRST can coexist on the same router. With the MGCP Gateway Fallback and SRST/E-SRST features running on the same router, both IP and analog phones can fall back to the SRST/E-SRST router when the WAN link to the Cisco Unified Communications Manager is down. Without the MGCP Gateway Fallback feature, FXO or PRI ports working under MGCP control from Cisco Unified Communications Manager are not available during fallback.
Q. If MGCP is enabled in a router with SRST/E-SRST active, will SRST and reliability, availability, and serviceability (RAS), admissions request (ARQ), and admission confirmation (ACF) gatekeeper services work?
A. Yes, the H.323 peer functions work for RAS, ARQ, and ACF; IP phones need to fail over to gatekeepers, and gatekeepers become the call-processing agent only for H.323.
Q. Is it possible to redirect a call to an external switchboard number through the PSTN when a call comes into the MGCP voice gateway running SRST/E-SRST but the extension is not registered on the router?
A. SRST/E-SRST does not provide any explicit functions to perform this redirection, but you can do it by using the normal Cisco IOS Software voice dial-peer and translation rule mechanisms. You can build a basic-telephone-service dial peer to match the direct-inward-dialing (DID) number you want to capture and then add a translation rule to prefix the PSTN access code, etc. The SRST
alias command lets you redirect calls to unregistered IP phones only in SRST mode. During MGCP operation, the basic-telephone-service dial peers are not used. If you do not know in advance whether or not the DID number will be resolved to a SRST phone and you want the call to go to the SRST phone if it does register, set the basic-telephone-service dial peer to preference 9 so that the SRST phone autogenerated dial peer (preference 0) is selected first if it exists.
Q. Does SRST/E-SRST support call forwarding on busy, all, and ring no answer (RNA)?
A. Yes, SRST supports these features using a global command
call-forward. Call forwarding on busy, all, and RNA are supported on a phone-by-phone basis in E-SRST mode only. You can also use the
alias command in SRST to allow calls destined for unregistered extensions to go to a designated extension.
Q. What call-transfer support does SRST/E-SRST offer?
A. Call transfer is supported when an incoming call is over the following interfaces and switch types:
● FXO and foreign-exchange-station (FXS) loopstart (analog)
● FXO and FXS groundstart (analog)
● basic-net3 and primary-net5 switch types
● Voice over Frame Relay (VoFR), voice over ATM (VoATM), and VoIP for Cisco gateway to Cisco gateway
● E&M (analog) and DID (analog)
● T1 channel associated signaling (CAS) with FXO or FXS groundstart signaling
● T1 CAS with E&M signaling
● All PRI and Basic Rate Interface (BRI) switch types
PSTN and ISDN BRI and PRI
Q. Are ISDN BRI and PRI supported in Cisco Unified SRST/E-SRST?
A. Yes, ISDN BRI and PRI interfaces are supported with Cisco Unified SRST/E-SRST.
Q. What switch types are supported on PRI and BRI interfaces?
A. All switch types are supported for PRI and BRI interfaces in Cisco Unified SRST/E-SRST.
Q. Can I use FXO ports in addition to T1 CAS ports for circuit connectivity to the PSTN with SRST/E-SRST?
A. Yes, you can mix and match different types of voice network modules.
Q. How many FXO ports can I assign for PSTN failover?
A. All the FXO ports on the router are available for PSTN failover.
Q. Are the commands default-destination and alias the same?
A. Both commands are global commands in
call-manager-fallback mode. The
default-destination command can forward incoming calls from FXO ports with an unknown or unroutable called number to an extension. If a default destination number is set, calls arriving on an FXO port are routed to the default destination number that is provided. If a default destination number is not set, the calls arriving on an FXO port receive a (secondary) dial tone.
alias command is more general and flexible, and removes any dependency on the FXO automatic private-line-automatic-ringdown (PLAR) mechanism that is associated with the existing
default-destination command. Also note that the
default-destination command does not work with the
connection-plar command. For example:
2600-srst(config-cm-fallback)# alias 1 50 to 5001
Calls to numbers in the 5000-5099 range that are not otherwise explicitly resolved to a specific extension are routed to the phone with extension 5001. This feature supports configurations in which only a subset of phones are supported in the
call-manager-fallback mode. Phone calls intended for phones that are not given fallback service can then be redirected to the specified extension.
Q. When both default-destination and connection-plar are configured on the SRST router, the call will not go to the default destination. Will default-destination override connection-plar?
A. When both commands are configured, the incoming call does not ring the extension number configured as
default-destination command does not override the
connection-plar command; you need to use the
Other Features Supported with SRST and E-SRST
Q. What codecs are supported in SRST/E-SRST?
A. SRST/E-SRST supports G.711 mu-law and G.729 on IP phones. You can specify the codec type in the dial-peer configuration on a Cisco Unified SRST/E-SRST router. G.711 mu-law is the default codec for local IP phone-to-IP phone calls and for IP phone-to-analog phone calls connected to the basic-telephone-service interfaces on the SRST router. G.729 is used by default for on-net calls through VoIP dial peers. SRST/E-SRST supports G.711 mu-law, a-law, G.729, and G.723.1.
Q. How do I configure a Cisco Unified SRST/E-SRST router to forward Dynamic Host Configuration Protocol (DHCP) requests to a DHCP server?
A. You should configure the SRST/E-SRST router as a DHCP relay agent to forward DHCP and BOOTP requests across the network or WAN. You should add the command
ip_address_of_dhcp_server under the router interface configuration portion and make sure that the
dhcp command is configured. Please note that the
service dhcp command is configured by default and it is not shown in the
show running-config command configuration. If it is explicitly turned off by using the no
dhcp command, you need to reenable it with service dhcp configured on the router.
Q. What will happen if the number of IP phones in the office exceeds the limit supported by the router?
A. The router rejects any phones trying to register after the limit is reached and does not service the phones.
Q. Is there a way to designate which phones get to register with the SRST/E-SRST router?
A. There are many ways to designate phones for registration. The easiest way is to configure different device pools on Cisco Unified Communications Manager - one for phones that you want to register to the SRST/E-SRST router after the fallback and a different one for phones that you do not want to register to the SRST/E-SRST router.
Q. When the number of IP phones in a branch office exceeds the maximum limit of supported phones, can I have two SRST routers in the same branch office?
A. Yes, you can group IP phones into different VLANs or subnets and allow each group to register with one of the SRST routers.
Q. What does dialplan-pattern do in SRST/E-SRST?
A. The dialplan-pattern command creates a global prefix that you can use to expand the abbreviated extension numbers (automatically obtained from the Cisco IP Phones) to expand into fully qualified E.164 numbers. The
dialplan-pattern command is also required to register the Cisco IP Phone lines with a gatekeeper. The
extension-length keyword enables the system to convert a full E.164 telephone number back to an extension number for the purposes of caller ID display, received, and missed call lists.
For example, a company uses extension number range 5000-5099 across several sites, with only the extensions 5000-5009 present on the local router. An incoming call from 5044 arrives from the company’s internal ISDN network, and this call includes the calling number as 4083335044 in its full E.164 format.
Q. What does the enhanced dial-plan pattern command do?
A. Cisco Unified SRST/E-SRST has keyword
extension-pattern to allow additional manipulation of the IP phone abbreviated extension number prefix digits. With this enhancement, the leading digits of
extension-pattern are stripped off and replaced by the corresponding leading digits of
dialplan-pattern. This process avoids a DID number such as 408-550-0001, resulting in a leading-0 4-digit extension number 0001.
Q. Does Cisco Unified SRST/E-SRST support Extension Mobility?
A. Cisco Unified SRST/E-SRST routers do not support Extension Mobility. IP phones rely on the support from Cisco Unified Communications Manager for Extension Mobility support. When in SRST mode, IP phones fall back to the SRST router with whatever profile it contains (extension#... etc.).
Q. Is interactive voice response (IVR) Automated Attendant supported on SRST/E-SRST?
A. Yes. IVR Automated Attendant is supported for Cisco Unified SRST/E-SRST. The sample script has added a routine to set up a call to the pilot number of the IP IVR on the Cisco Unified Communications Manager site, and if the call fails the script continues running on the local SRST/E-SRST router.
Q. How does it work?
A. When an incoming call to the IVR Automated-Attendant pilot number arrives on the router, the router tries to set up a call leg to reach the Cisco Unified Communications Manager pilot number or IP IVR number. If Cisco Unified Communications Manager is up and the IP IVR port is available, the call succeeds and the calling party hears the prompts played by the IP IVR. However, if the WAN or Cisco Unified Communications Manager is down or the IP IVR port is busy, the call setup fails, and the calling party hears the prompts played by the SRST/E-SRST router. Note that the calling party must dial the IVR Automated-Attendant number of the SRST/E-SRST router because dialing the Cisco Unified Communications Manager pilot number (IP IVR number) does not invoke the IVR Automated-Attendant feature on the SRST/E-SRST router.
Q. Can I dial a different phone number to make a call by using an IVR Automated-Attendant script on the SRST/E-SRST router?
A. No, in order to invoke the IVR Automated-Attendant feature on the SRST/E-SRST router, you need to call the IVR Automated-Attendant pilot number. Calling other numbers will not invoke the IVR Automated-Attendant feature.
Q. Where can I get more information and sample configurations for SRST and E-SRST?