IPT Features Query(Call time limit, dedicated lines, Off hook dial tone)
Please let me know is it possible to have below features
1.Call time limit( after 20/30 min calls should disconnect automatically)
2.dedicated lines for executives(How do we dedicate 3 to 5 SIP channel from 30 channel from PSTN SIP trunk for executive/minister in case of peak calls)
3.Off hook dial tone( SIP trunk can handle 30 calls simultaneous, if all the 30 channels are occupied and after that any user wants to dial PSTN via SIP trunk can we configure to let users know that all SIP channel are busy as soon as the user OffHook)
There's some TCL scripts floating around that can limit the call duration on CME/CUBE. CUCM has a "Maximumm Call Duration Timer" service parameter.
Depending on your SIP carrier, you may be able to do MLPP over SIP to disconnect existing calls if an executive needs to dial out- http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/cmfsgd701/fsmlpp.html
Probably easier to just give them a dedicated POTS line each or split the channels into 2 SIP trunks if possible.
SIP doesn't have a method to inform in real-time if all your active connections are used up without actually sending a call. You could potentially build some sort of custom script to check how many calls are active and use PLAR on all the phones to send them to somewhere that can read a message or play a tone based on how many connections are being used.
But the CUCM service parameter is global and affect every single call in CUCM, so no.
The only option, for any system, would be TCL scripting.
Also no way to provide a message as soon as they pick up the phone, even with a PLAR pointing at some place, it would follow that behavior even if there are channels available, and it also would need to follow the CUCM timers if added to their regular CSS.
#2 as mentioned, MLPP
I would suggest you to reach a partner with the list of requirements your company has, so they can evaluate what system would be best for you.
He didn't say he wanted the timer on a per-call basis but you're probably right in that's what he wanted :)
For playing the message, you would pretty much need to setup a system to PLAR to a 3rd party device and then have it play a message or not based on available channels and then collect DTMF digits in order to place the outgoing call if channels are available based on checking the number of active calls on the CUBE or some sort of API the SIP provider offers. Definitely not anywhere near built-in and would require hiring an integration partner.
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
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Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...