ASK THE EXPERTS: Cisco Unified Border Element for PSTN SIP trunks

[ciscomoderator]

With Randy Wu

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn best practices on how to configure and troubleshoot Cisco Unified Border Element (UBE) for the public switched telephone network (PSTN) Session Initiation Protocol (SIP) trunks with  Cisco subject matter expert Randy Wu. Randy Wu is a senior customer support engineer in the Multiservice Voice team at Cisco in Sydney. He has vast experience and knowledge configuring, troubleshooting, and designing Cisco UBE, gateways, and gatekeepers, working with H323, MGCP, and SIP protocols. He joined Cisco as a systems engineer in 1999. He holds CCIE certification (#8550) in Service Provider, Routing, and Switching and Voice.

Remember to use the rating system to let Randy know if you have received an adequate response.  

Randy might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the IP Telephony discussion forum shortly after the event. This event lasts through August 5, 2011. Visit this forum often to view responses to your questions and the questions of other community members.

[yuanwu]

Hi, Jose

thanks for the information.

1. the 15.2.1T IOS for other platform will be there soon.

2. this feature was designed to improve the inter-op issue with CUCM when hold and resume, the message for hold and resume will be handled locally at CUBE without passing through to the provider.

Rgds/Randy

[JULIUS KINSLER]

I understand you need to purchase licenses for your concurrent sessions. This may be a basic questions but I came across this discussion as I was looking for the answer. How can you verify how many concurrent sessions you have installed on a SIP gateway? I am working with a 3900 ISR G2 If it makes a difference. Also, Is there a command I can issue to view how many concurrent calls are being used?

Thanks,

Julius

[yuanwu]

Hi, Julius

thanks for your question.

The session license is not enforced in the software control yet, it is based on your network scale and design, then you need to purchase the accordingly number of the sessions which will be used, also you can buy the bundle option which includes the maximum sessions support in the platform you bought.

please check the following URL for more details,

http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html

From the ISR G2, you can use "show call active voice brief", "show call active voice compact" to have an idea of concurrent calls being made.

Rgds/Randy

[jonathan.akins]

Randy - We are evaluating using the CUBE as the PSTN SIP provider SBC for MS Lync and also a gateway for terminating analog lines (fax machine) in small offices.  Have you seen this configuration before and if so, any problems?  Thanks.

[yuanwu]

Hi, Jonathan

thanks for your question.

The CUBE is running on ISR like 28/3800, ISR G2 29/3900 etc platform,which can be TDM GW with CUBE function at the same time,  you can have the FXS/FXO modules in the same box for the analog fax or phone connection.

Rgds/Randy

[Lin Zhu]

hi,

nice to meet you. I am studying voice now, I got a confused question , that is : I can not understand the relationship of many different users in call manager, such as relationship of Phone, DN, enduser, application user, etc.

do you have some documations regarding this topic?

thanks