With Justin Dew
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about integrating Cisco Unity Connection into your environment and troubleshooting the many features that are available with the Cisco Unity Connection voicemail solution .
Cisco Unity Connection is a voicemail and unified messaging platform with a comprehensive feature set and is based on the Linux unified communications operating system. With Cisco Unity Connection, you can access and manage voice messages in variety of ways, using your email inbox, web browser, the Cisco Jabber messaging integration platform, Cisco Unified IP Phone, smartphone, tablet, and more. Cisco expert Justin Dew will cover and answer all of your questions about Cisco Unity Connection.
Justin Dew is currently a member of the Applications TAC team in RTP NC. His area of expertise is in Cisco Unity connection and Cisco Unity. Justin has had six years of experience troubleshooting and working with voice products, including unified communications, Cisco Unity, presence, and Jabber in the TAC organization. He graduated with a bachelor's degree in electrical and computer engineering from Georgia Tech and has achieved certifications in CCVP and as a Domino administrator and Cisco Unity specialist.
Remember to use the rating system to let Justin know if you have received an adequate response.
Justin might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation in Collaboration, Voice and Video, sub-community, IP Telephony discussion forum shortly after the event. This event lasts through December 20, 2013. Visit this forum often to view responses to your questions and the questions of other community members.
Typology aside what is the call control integration? SIP dial-peers? MWI not supported? There are no details on how the integration works, supported topologies doesn't really clarify the best practice approach for integration.
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Call control is performed by the CUCME at the remote site or the SRST ccm server that is setup at the remote site. The remote site UC server can be setup with either sip or sccp call control.
I mainly deal with the UC side setup of this feature so I am not as familar with the actually configuration required on the CUCME side and the SRST side. As for a sip integration I suspect a SIP dial peer should be sufficient to route the srst phones over the the UC srsv server.
You are correct MWI is not currently supported with SRSV for UC.
I am using unity connection 8.5 in our company , i am have an issue with one of my extension where user has cheked all voice mail but Headset red LED still blinking.
You can verify your setup as per the following link