Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to get an update on Cisco Unified Communication Express products with Cisco expert Tony Huynh. Tony is a technical marketing engineer for Cisco Callmanager Express (CME) at Cisco Systems, Inc. He is a (CCIE # 11056) Cisco Certified Internetwork Expert in routing, switching and voice with eight years of design and support experience in the information technology industry. Over the years, he has worked for various corporations including several Fortune 500 companies. Tony is an expert in documenting, designing, and implementing various communication systems. His areas of expertise include technologies such as routing & switching as well as IP telephony.
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We have CUCME 7.0 installed in our lab on an 1861 router and it works great. However, we are having an issue getting the Live Record feature to work correctly. To be specific, it works, but we are unable to retrieve the recording, as they are not in the personal mailboxes. We have no earthly idea where to retrieve them from. The documentation states that the recorded call will show up in the personal mailbox of the person who initiated the recording, but its not there, and we get no MWI. The Unity Express version i believe is 3.0. Does it need to be Unity Express 3.2, and, if so, is there a CUE 3.2 available for the 1861 platform, as we have been unable to locate it.
Any info you can provide is greatly appreciated.
Thanks in advance
Did you configure hardware conferencing on the system? Also, have you configured the live-record softkey on CME and CUE?
Thanks for your reply. To answer your questions, yes hardware conferecing has been configured, the live record softkey is there, and the live record pilot has been configured in CUE. When we press live record, it works as designed, I hear the beeps, and see that the call has been sent to conference w/the live record pilot number. We just dont know where to retrieve the recording as it is in neither the initiators mailbox or the mailbox of the user on the other end.
The recording should be stored in the voicemail box of the person that hit the live record softkey. If this isn't working, then it appears to be a bug.
The fact that you hear the beep tells me that the CME is correctly starting a hardware conferencing to conferencing in the live record pilot with the call.
Do you have a TAC case open here that I can check?
Our CAM from cisco had one open, but I dont have the number handy. On the 1861 router we have CUCME 7 but CUE 3.0. The documentation we have references CUE 3.2. However, the engineer we were working with said that there is no CUE 3.2 for the 1861 platform, as its basically the same platform as the UC500. We also looked on CCO and couldnt find the image for CUE 3.2 for 1861/UC500 platform. So we are going to try it on a 2800 and see what happens.
Hi Tony, I have a CCME 4.0 and it is working fine and we planning to update to the latest CCME version. I'm wondering if the latest version will have features like Force Authorization Codes and a way to obtain call reports like CUCM. From the point of view of the users, these are important features.
We don't have forced authorization code yet on the latest CME version. We have a FAC script, but it is not TAC supported. We are looking to add this feature in a future release of CME.
I see an option for FAC under Telephony Service in CME 7, but I cannot find tech info for that... So, if it is not for forced authorization codes, then what it is for?
In case you are interested, you can find the forced authorization code script here.
I have a Cisco 2851 with IOS 12.4(11)XW9 and CME Version 4.2
This CME receives calls from two different H.323 peers.
172.x.x.x and 10.x.x.x
Outgoing calls work ok to both ip destinations, but incoming calls from 10.x.x.x do not work. They hit the dial-peer 100, I need them to hit dial-peer 150.
If I specify a phone number with "incoming called-number" a call to this number will work from 10.x.x.x, but that is not a solution.
The phones at this site are numbered 15xxx
The two dial-peers are configured like this:
dial-peer voice 100 voip
session target ipv4:172.x.x.x
incoming called-number .
ip qos dscp cs5 media
dial-peer voice 150 voip
voice-class codec 1
voice-class h323 1
session target ipv4:10.x.x.x
incoming called-number .
ip qos dscp cs5 media
You could try making the incoming called-number pattern more specific on dial-peer 150. Otherwise, delete dial-peer 100 and re-add it back in -- it will be added lower in the running config and thus dial-peer 150 will be matched first.
Users from both locations need to dial the same numbers, so I need the same incoming called-number in both dial peers.
Your second suggestion would make calls from session target ipv4:10.x.x.x work, but I guess calls from ipv4:172.x.x.x would stop working.
What I need would be a command like "incoming calling-number", or to make the CME match incoming calls based on originating ip address.
A config example of three CCME all making calls to eachother would help a lot.
You could use the "answer-address ANI_STRING" variable underneath the dial-peer to match the calling number.
More info here:
Our company is trying to harness the reporting information from ICM Router Log Viewer.
Can you tell me where that information is pulled from? I have looked through the ICM 7.1 DB schema, but I cannot find an entry referring to it.
I want to take the data and create a feature that will notify engineers when errors occur.
I can try and research this for you, but this wouldn't apply to CME. CME exports CDR to either syslog, radius or a FTP server.
In a non-autoanswer environment, should there be any issues with having the ACD line set with a Busy Trigger > 1, as long as it is set less than Max Calls?
i am a beginner in VOIP network and i have question, if we have 2 gateway and connected by WAN connection and each one is connected with group of ip phone and i apply SIP signaling at this gateway what is the procedure when we make call (ip phone is SCCP and gateway is SIP)?
Sorry if it trivial question.......
The signalling would be SCCP between the phones and CME and then between CMEs would be SIP (if you configure a SIP dial-peer). If you configure a regular dial-peer and DON'T specify session protocol SIP, then the default protocol (H323) will be used to negotiate the call between the 2 gateway systems.
i have a customer has cuple of issues
first there is a problem with timestamp of CUE on top of CME
i would sugessted them to make CME as ntp server and point CUE to cme for ntp and chose the right time zone as well ?? is that only required for CUE
also they have one phone unable to change ring ton
they reset all to factory defualt all phones work normally except that unit ? using CME
also if i wanna let a phone has multiple channel to answer more than a call
is it better to configure more than one DN with the same number and add then in one button with overload line (lo) regarding the version have no support for new line feature called oct-line ( i am i right in this )
thank you very much
Setting the NTP server to CME on CUE is the correct thing to do. Are they trying to change the ringtone to a distinctive ring tone or one of the ones that comes on the phone?
With CME 7.0, each ephone-dn can have up to 8 channels, thus can have up to 8 calls. If you want to share a number across multiple phones, you can configure overlay dns.
With overlay lines (with the same number), you get the following behavior. Lets say number 101 is assigned to multiple dns that are overlayed across 4 phones. The first call into the system for number 101 would ring all 4 phones and the first to answer gets the call. The next call that comes into the system for 101 rings the 3 remaining non-busy phones and so on.
ok first thanks for ur answer
in regard to ring ton, it is just the defualt ring tons !!
for overlayed lines
based on the way u described i think this is shared line not overlayed
if i configure overlay line for the same number in the same phone
this way the caller can recieve more calls on the same line on the saem phone each DN is dual-line thus he/she can hold forward each call at the same time
i think as u mentioned if i configure severla DNs with the same like lets say 101
and i make 101 on three phones the call allocation will be based on DN prefrence lower prefrence asnwered first
but if phone one answered
then second call come will go to second phone
the ephone should look like
if i want it like the first call answered and we need to let the second call ring the second phon and show callwaiting in the first phone the busy one
it should be configured like
am i correct ?
by the way i love to know about the ring ton issue if you have any details link
thank you very much
Apologies for the mix up. Here is the link.
are there any cisco software tools that simplify the job of setting ip a
sip trunk on cme? does the sip trunk *have* to be done manually in the cli?
I have cme 4.1, but the web gui doesn't include sip setup. Thanks for
You can try Cisco Configuration Professional - I'm not sure if it added support for SIP trunks yet. Otherwise, you could use the following CME SIP Trunk Deployment Guide.