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Unity connection SRSV

Matt Smith
Level 1
Level 1

Hello Unity wizards;

 

I've been following the guide here: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/srsv/guide/9xcucsrsvx/9xcucsrsv010.html

Software Versions:

Cisco Unity Connection version: 9.1.1.10000-32      

Cisco Unity Connection SRSV version: 9.1.1.10000-32 

Cisco Unified CM Administration: 9.1.1.10000-11

CUBE: C2951-UNIVERSALK9-M), Version 15.2(4)M2,

 

SRSV contains these major steps which I have completed:

Workflow in Cisco Unity Connection SRSV

1. The administrator installs Cisco Unity Connection on SRE-900/SRE-910 series blade or MCS 7845/MCS 7825. For more information refer to

2. Connection starts in the Demo mode. Run the CLI command utils cuc activate CUSRSV to convert standalone Connection server to Connection SRSV server. For more information on installation of Connection SRSV, refer to the "Overview of Cisco Unity Connection SRSV in Connection 9.1(1) and Later" chapter of this guide.

3. The Connection SRSV server disables some of the Connection components and displays only the following Connection components within the Connection SRSV Administration:

–Users, with the list of administrators and subscribers of the branch. For more information on the user settings of Connection SRSV, refer to the "Cisco Unity Connection SRSV Administration - User Settings Interface" chapter of this guide.

–Templates, with only the Call Handler templates. For more information on the template settings of Connection SRSV, refer to the "Cisco Unity Connection SRSV Administration - Template Settings Interface" chapter of this guide.

–Distribution Lists, with only System Distribution Lists. For more information on the distribution lists, refer to the "Cisco Unity Connection SRSV Administration - Template Settings Interface" chapter of this guide.

–Call Management, with only System Call Handlers and Directory Handlers. For more information on call management for Connection SRSV, refer to the "Cisco Unity Connection SRSV Administration - Call Management Settings Interface" chapter of this guide.

–Networking, with central server configuration. For more information on the user settings of Connection SRSV, refer to the "Cisco Unity Connection SRSV Administration - Networking Settings Interface" chapter of this guide.

–System Settings, with only Schedules, Conversations, Enterprise Parameters, Plugins. For more information on the user settings of Connection SRSV, refer to the "Cisco Unity Connection SRSV Administration - System Settings Interface" chapter of this guide.

–Telephony Integrations, with only Phone System, Port Group, Port, Security. For more information on telephony integrations, refer to the "Cisco Unity Connection SRSV Administration - Telephony Integration Settings Interface" chapter of this guide.

–Tools, with only Custom Keypad Mapping. For more information on the user settings of Connection SRSV, refer to the "Cisco Unity Connection SRSV Tool Settings" chapter of this guide.

4. The administrator logs into the Connection Administration page and navigates to the Branch Management page. For more information on how to configure the central Connection server for Connection SRSV, refer to the "Configuring Cisco Unity Connection SRSV Settings in Cisco Unity Connection 9.1(1) and Later" chapter of this guide.

5. The administrator enters the Fully Qualified Domain Name (FQDN), administrator username, and password for the branch connection node. Connection and Connection SRSV verifies registration and associates the branch to the Connection server. For more information on how to configure the central Connection server for Connection SRSV, refer to the "Configuring Cisco Unity Connection SRSV Settings in Cisco Unity Connection 9.1(1) and Later" chapter of this guide.

6. You must set a method to provision the users from the central Connection server to the branch system. For more information refer to the "Configuring Cisco Unity Connection SRSV Settings in Cisco Unity Connection 9.1(1) and Later" chapter of this guide.

7. The administrator imports the subscribers by searching on users details, such as extension/phone number, already existing in Connection and selects those users. For more information on how to configure the central Connection server for Connection SRSV, refer to the "Configuring Cisco Unity Connection SRSV Settings in Cisco Unity Connection 9.1(1) and Later" chapter of this guide.

8. The administrator selects the "Sync Provisioning" button to push the subscribers to Connection SRSV. The provisioned status is displayed on the Connection SRSV Administration page. For more information on how to configure the central Connection server for Connection SRSV, refer to the "Configuring Cisco Unity Connection SRSV Settings in Cisco Unity Connection 9.1(1) and Later" chapter of this guide.

 

After completing these steps the voice ports from SRSV are registered on CUCM.

 

What I seem to be missing is the glue that binds SRST on the CUBE to SRSV during failover. I haven't found documentation on gateway configuration for Unity Connection SRSV. It becomes more challenging when there are two SRSV products and most of the documentation or posts is in relation to the older EOL'd product which required a unified messaging service.

Where I'm at right now:

When I disrupt network connectivity to CUCM, I get ephone registration for all my phones but I don't see any SRSV ephones registered. I'm assuming the integration with CUCM allows SRSV to register its ports as ephones during a failover scenario, but if this isn't the case please let me know.

Any help is appreciated, TAC has thrown me between reps for the last 3 weeks without a resolution, and the demo period for this module is almost up, and more importantly the production requirement for WAN failure voicemail services at a branch office is rapidly approaching.

Normal call flow:

Inbound call -> CUBE -> CUCM -> Unity IVR -> ext/hunt group

Failover call flow:

Inbound call -> CUBE -> Call tries to hairpin back out ITSP trunk as there is no applicable dial peer.

CUBE configs:

Inbound dial-peer:

dial-peer voice 10 voip
 description ** Incoming call from trunk main line/main line TF **
 call-block translation-profile incoming all-blacklist
 call-block disconnect-cause incoming call-reject
 destination-pattern MYDID
 monitor probe icmp-ping 10.0.6.30
 session protocol sipv2
 session target ipv4:10.0.6.30 <-- CUCM
 session transport udp
 incoming called-number MYDID
 voice-class codec 1  
 dtmf-relay rtp-nte
 dtmf-interworking standard
 no vad

Outbound dial-peer:

dial-peer voice 20 voip
 description ** Outbound call to trunk **
 preference 4
 destination-pattern ....T
 session protocol sipv2
 session target ipv4:xxx.xxx.xxx.xxx
 session transport udp
 voice-class codec 1  
 dtmf-relay rtp-nte
 dtmf-interworking standard
 no vad

 

 

call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 10.0.250.4 port 2000
 max-ephones 150
 max-dn 500
 system message primary SRST:Phonesys in Failover mode
 keepalive 25
 voicemail 3000
 call-forward busy 3000
 call-forward noan 3000 timeout 20
 moh "flash:/SampleAudioSource.ulaw.wav"
 time-zone 12

 

Unity Connection SRSV System Call Handler "Opening Greeting" extension is set to 3000.

 

Thanks,

Steve

2 Replies 2

Mike Lydick
Level 1
Level 1

I used a SIP integration with SRSV and CUCM to avoid having to do a SCCP/SRST configuration for the voicemail ports. Built a parallel integration with CME/SRST (SIP dial-peer and a entry in SRSV to allow connection from the SIP bound interface). You are using Callmanager Fallback but could do similar. 

If you need more details let us know.

 

 

Mike

 

 

Hi Mike thanks for the response.

I ended up going with SIP between the CUBE/Unity SRSV as well with dial peers for DID's (higher preference ) and the voicemail extension.

I ran into an issue where calls from Unity SRSV to registered ephones were going back out the ITSP SIP trunk. The following setting resolved it:

voice service voip

no supplementary-service sip refer

From my understanding the CUBE would attempt SIP refer first, and when refer is not an available option it will drop back to SCCP which then uses the ephone related dial peers.

Hopefully this helps someone else as well.