ASK THE EXPERT - UNIFIED COMMUNICATIONS EXPRESS

Unanswered Question
Jan 5th, 2009

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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Kenneth Mohammed Mon, 01/05/2009 - 09:46

Hello Tony,

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

Tony Huynh Mon, 01/05/2009 - 13:34

Did you configure hardware conferencing on the system? Also, have you configured the live-record softkey on CME and CUE?

Kenneth Mohammed Mon, 01/05/2009 - 13:47

Tony,

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.

Tony Huynh Mon, 01/05/2009 - 15:41

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?

Kenneth Mohammed Mon, 01/05/2009 - 16:09

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.

Tony Huynh Mon, 01/05/2009 - 16:48

CUE 3.2 isn't available as software pack yet for the UC500 or 1861. Let me know how it goes with the 2800.

david-lima Mon, 01/05/2009 - 10:35

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.

Thanks

David.

Tony Huynh Mon, 01/05/2009 - 13:35

Hi,

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.

jbarcena Wed, 01/07/2009 - 15:01

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?

Tony Huynh Wed, 01/07/2009 - 16:30

The FAC under telephony-service is feature access code and NOT forced authorization code.

vigleik Tue, 01/06/2009 - 05:07

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

preference 1

destination-pattern 10...

session target ipv4:172.x.x.x

incoming called-number .

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

no vad

dial-peer voice 150 voip

preference 1

destination-pattern 11...

voice-class codec 1

voice-class h323 1

session target ipv4:10.x.x.x

incoming called-number .

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

no vad

Tony Huynh Tue, 01/06/2009 - 08:43

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.

vigleik Wed, 01/07/2009 - 00:12

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.

Vigleik

Matthew Berry Tue, 01/06/2009 - 05:08

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.

Tony Huynh Tue, 01/06/2009 - 09:10

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.

jpsweeney77 Tue, 01/06/2009 - 08:41

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?

titomontos Tue, 01/06/2009 - 15:35

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.......

Tony Huynh Tue, 01/06/2009 - 15:40

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.

Marwan ALshawi Wed, 01/07/2009 - 17:21

hi toy

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

Tony Huynh Wed, 01/07/2009 - 17:45

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.

Marwan ALshawi Wed, 01/07/2009 - 18:48

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

ephone 1

button 1o1,2

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

ephone 1

button 1c1,2

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

Tony Huynh Thu, 01/08/2009 - 08:43

Yes, you are correct in your statements above. Here is a link that helps with ringtones.

Thanks

Tony

tonyspcrepairs Tue, 01/06/2009 - 17:23

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

any pointers.

tonyspcrepairs Tue, 01/06/2009 - 18:09

thanks for your response. I downloaded CCP only to discover it doesn't support my 2651XM router.

I've also seen the document you mentioned but none of the settings work with my sip provider, and my sip provider doesn't know about cisco routers with cme 4.1

Do you know of any other config examples that are known to work with standard sip providers? thanks for any further advice.

tonyspcrepairs Wed, 01/07/2009 - 05:00

tescointernetphone.com

sip-server: sip.tescointernetphone.com

regisrar server: sip.tescointernetphone.com

somuchoton Wed, 01/07/2009 - 01:10

We WWIL are going to start Broadband services very soon. And here we are using Cisco 7206 VXR router, Core switch Cisco WS-C3524-XL, Cisco manageable L3 switch. I just want to get some technically trained about the physical connectivity for broadband services with all the devices connected and the other servers. I just also want to know how to configure all the required devices for broadband services. Before all these aspects please tell me how I will provide a customer with Broadband services as well as with a live video and voice over the single fibre by WAN Technology. For this technology what are the devices we require and what will be the physical connection. So please help me out as I am very new to this sector.

Tony Huynh Wed, 01/07/2009 - 16:31

Hi, please see thread below. There isn't just a single set of hardware that you need. It depends on what type of deployment you are looking for.

Eg. A large headquarters type of deployment would require the larger Catalyst switches and then depending on how much bandwidth you need, you can choose the appropriate router. Please also keep in mind the type of PSTN and WAN trunks you need.

Javier Perez Lledo Wed, 01/07/2009 - 01:32

Hi,

I would like to know how to provide a Single site CME express with redundancy. The only way I know how to do it is:

- Dual hw/sw (+ manual phone lines switch over)

- Clone CFG's: ephone & DNs configurations on each router

- Dual TFTP entry on DHCP server ( 1 per CME)

- Doble licensing

However on the SRND I found page 2-1:

" The single-site model has the following design characteristics:

- Single Cisco Unified CallManager Express (Cisco Unified CME) router or dual Cisco Unified CME router for redundancy

- Maximum ..."

Could you give me (us) more details what the "dual Cisco Unified CME router" means. I dont find any reference to this term on the CCO.

Thanks in advance.

somuchoton Wed, 01/07/2009 - 01:37

Actuall i want to know the hardware devices required for live Video and Voice over WAN technology with our Broadband Services

Tony Huynh Wed, 01/07/2009 - 15:46

Hi,

All this depends on a few things:

(1) Size of your office

(2) type of phones you wish to use

(3) types of services you need (conferencing, etc)

This will help determine what type of router and switch you need. It will also help determine which type of phones you should deploy.

tonyspcrepairs Wed, 01/07/2009 - 15:06

hi Tony I managed to get my sip trunk working by some flukey miracle. From a pstn phone I can dial my sip number and my ip phone rings. But I don't know how to dial out - don't know what button I should press to specify an outside sip call. Right now f I try to call a pstn number I get 'unkown number'. thanks for any pointers.

Tony Huynh Wed, 01/07/2009 - 15:44

Did you create a VOIP dial-peer to point out to the PSTN via the SIP trunk.

Something similar to:

dial-peer voice 3 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 9[2-9]..[2-9]......

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 4 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 9[0-1][2-9]..[2-9]......

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 5 voip

description **911 Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 911

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 6 voip

description **Emergency Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 9911

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 7 voip

description **911/411 Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 9[2-9]11

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 8 voip

description **International Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern 9011T

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

!

!

dial-peer voice 9 voip

description **Star Code to SIP Trunk**

destination-pattern *..

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

tonyspcrepairs Thu, 01/08/2009 - 13:42

I have created the dial-peers now in my config as per your instructions and there is a slight improvement but I still can't get a successful line out. I think one of the problems is that the dial-peers given cater for the american telephone system and I'm in the UK. If I press 9 and then dial the full UK number I get a message saying that the number I dialed 'is not connected'. Same if I press *. I've got 11 dots after the star and the 9 to represent the full UK phone number but I can't tell what the phone is actually dialing. Are there any configs that cater for UK based numbers?

Tony Huynh Thu, 01/08/2009 - 14:37

One thing you need to keep in mind is that because you are using a VOIP dial-peer (for SIP trunk) to send the call out to the PSTN, the CME will not strip the digit "9". Thus 9 is probably getting prepended to your outbound dialed number. You would need to configure a translation-pattern to strip the 9 and then send the rest of the digits outbound.

tonyspcrepairs Thu, 01/08/2009 - 14:59

I've been searching on google but I couldn't find any config examples showing such a pattern Do you know where I can find a transfer-pattern that would strip a digit as you say?

tonyspcrepairs Sat, 01/10/2009 - 05:47

ah yes I see it now, thanks for that Tony. At the moment I'm using just a string of 11 dots to represent a standard UK phone number like this:

dial-peer voice 2 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

destination-pattern ...........

it works but I expect this is not the right way to do it.

tonyspcrepairs Thu, 01/08/2009 - 15:57

I changed the destination-pattern 9 to 0 and I found some info after a bit more searching and I tried:

translation-rule 100

Rule 1 ^0........... 0

but that didn't work.

I also tried:

translation-rule 100

Rule 1 ^0.* 0

but that didn't work either.

I made sure there was 'translate-outgoing calling 100' under the dial-peer. Can you see where I might be going wrong? I still can't dial out.

stevenle Wed, 01/07/2009 - 15:53

Hi Tony,

Happy New Year!

I have Unity version 4.1 and am attempting to configure the Live record feature. However when I go to create the routing rule there is one already in place. But all the fields are grayed out and it will not let me edit. How can I get this routing rule to allow me to change it?

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