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Feature and design question for CME and CUE.

Bernard Magny
Level 1
Level 1

I have recently installed and now using a CME and CUE system. I have a few question about some requested features the users are interested in.

First a few technical issues. I have a 32 channel DSP, my telephone provider gives me access to a ISDN 30 (30 lines) and I also have a 4 way voice FXS module. The issue is, when I enable all 30 channel of my ISDN using :

controller E1 0/0/0
pri-group timeslots 1-30

The command gets deleted because the 4 way FXS module takes 4 channels and there is not enough DSP channel left for the ISDN.
I only use 2 of the 4 ports on the FXS, so I like to disable 2 ports, but I can’t seem to find a way.

If I only enable 25 channels of my E1 controller it then works, but does this mean I could be missing phone calls? If someone comes in the ISDN on one of the channel not assigned to any controllers?

Other issues, less hardware oriented is the following:

1. I tried to create a “group” mailbox on the unity system. It seems to work, group members have access via a new option when they go in their mailbox. The problem, is that I can’t get the auto attendant to transfer to this mailbox. You see I use a parallel hunt-group and would like to transfer the call to that group mailbox if no one answers. I don’t know how to do that.

2. Users, on the old system, use to have short cuts. Basically, users could dial 9+ext of a known co-worker and it would dial the mobile number of the co-worker. I had to program all those “short codes” in the system before with the mobile numbers. I like to do this again, perhaps with a translation rule, I just not sure how to do it. I am limited to 15 rules so I can’t put in all the mobile numbers like that, the best would be via a xml file or something easy to change like that. Any ideas?

3. I have several users that have both a desk phone, normally a 7911G, but also have a software phone at home via VPN. Best would be so that the extension for the desk phone and the remote software phone be the same. So if someone in the office dials ext 513 both the 7911 and the software phone in the remote location rings. Maybe that is not such a great idea, not sure, but there is also the issue that even if it’s a different ext number for the remote phone, I can’t apply the same username to this new ephone. Does this mean each users that have different locations and different phones needs different user names? Basically what would be the best practice setup for users with multiple phones but always only one active?

4. On the CME web access, how do I log out so I can log in as a different users? It seems like IE has remember my login details and no matter what I try with cookies or password, I can make it forget. Also the logout button on the CCME.html site only closes the browser, not logs out of the application.

5. I don’t seem to be able to remove a MAC address from a deleted  ephone, so I am force to skip numbers. How do I change the MAC address of an ephone in the configuration of CME?


Well this is about it, I am sure as the system is being use, I will have more issues from the users. Thank you in advance to anyone that can answer part of my questions and sorry again for the wall of text.

14 Replies 14

Steven Holl
Cisco Employee
Cisco Employee

Analog ports require DSPs for signaling detection so DSPs are allocated at boot for the ports in the voice card, regardless of whether the ports are shutdown or not.  You can either purchase more DSPs, provision a fractional T1 (>=28 channels, though if you do this, the provider needs to reflect this as well), or pull out the 4 port card and get a 2 port card.

1. For the GDM, you need to get the call to be directed to VM with a diversion number of the GDM number.  The easiest way to do that is to have an ephone DN with the GDM number assigned, and CFB/CFNA/CFA set to VM.  Then have the AA transfer to the GDM number which will forward to VM with the diversion for the GDM inbox number.

2. I think the best way for your mobile numbers are to either set up global speed dials with XML:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmespeed.html#wp1033709

Or use num-exp:

http://www.cisco.com/en/US/partner/docs/ios/12_3/vvf_c/dial_peer/dp_confg.html

3. You can't use the same username, but that's not a big deal.  Just don't assign a username to the remote phone.  Map the office line as a shared line to the remote phone.  That's it.

4. Agreed.  The CME GUI doesn't handle this well.  Delete cookies and clear browser cache, or user a different browser for the other user.

5. Do this:

ephone 5

no mac-address

mac-address

Thank you. I will try those solutions and report back my progress.

Hello
again,

As soon as I create a ephone-dn for the GDM, my parallel hunt group does not work
anymore. All calls goes to that ephone-dn and if there is no CFB on it, it just sounds busy since there is no phone assign to it.

This is what I tried that did not work:

ephone-dn 298

number 500

description GDM for group 500

call-forward busy 100

call-forward noan 100 timeout 300

voice hunt-group 5 parallel

list 512,513,514,516,517,518
pilot 500

Thanks



You can't have the pilot number and the GDM number be the same.  Either you need to change the pilot number (the easy fix), or you need to change the GDM number in CME and reflect that new number in CUE for the GDM mailbox number.

Ok, I can easily change the pilot number. The problem, how do link the hunt group with the GDM. Currently the AA transfer calls to the pilot number 500. If no one answer, it needs to go to a GDM.

Thanks.

Oh you need to translate the redirecting number to do that.  Apply a translation profile on the dial-peer to CUE to translate the redirect-called number from the pilot number to the GDM number.

I am not 100% following you. I’m pretty new at this, so please bear with me.

So I now have a hunt group with a pilot number 501. I have a ephone-dn with the number 500. I Also have a GDM on the CUE with number 500.

Since there is no ephone connected to the ephone-dn, I forward all calls to the hunt group. Anything else, it just goes to the GDM mailbox.

Here is my config:

voice hunt-group 5 parallel

list 512,513,514,516,517,518

pilot 501

ephone-dn  298

number 500

description GDM for group 501

call-forward all 501

So I don’t see how it would reach the GDM. I put in a translation rule as you suggested, but it did not help.

voice translation-rule 4

rule 1 /501/ /500/

!

voice translation-profile GDM

translate redirect-called 4

dial-peer voice 100 voip

translation-profile incoming GDM

destination-pattern 10.

session protocol sipv2

session target ipv4:10.10.100.208

dtmf-relay sip-notify

codec g711ulaw

no vad

I know I am missing a few things here, or even some fundamentals, but as I said, I am pretty new at the voice thing.

Thanks

Bernard,

You're close.  Since the CUE dial-peer is an outbound dial-peer match, the translation profile needs to get applied outbound.

Change to:

dial-peer voice 100 voip
no translation-profile incoming GDM
translation-profile outgoing GDM

Also, let's clean the translation rule up a little while we're at it, and make it so that only 501 gets translated.  The current rule would match 5010->5000 and 1501->1500 since you didn't define start and end of string.  Use:

voice translation-rule 4
rule 1 /^501$/ /500/

Yes, I actually realised the outgoing just before you answer. But, it still does not work.

I don’t understand how a call would make it to the CUE.

A caller gets transfer from the AA to the number 500. 500 is set on forward all to hunt group 501. But then, how would it know to go on voice mail if 501 does not pick up or is busy ?

Thanks again.

voice hunt-group 5 parallel

list 512,513,514,516,517,518

pilot 501

final 500

ephone-dn  298

number 500

description GDM for group 501

call-forward all

call-forward busy

It still not working. I am having an issue troubleshooting it. I don't know what debug to use to see the call flow, cause it just does not work.

First example:

Currently, a call comes in to the AA. The user select the hunt group option, that works, it gets to group 501.

No one answers hunt group 501. The call is simply dropped.

Second Example:

User call directly to hunt group 501.

No one answers, it goes to a fast busy sound.

Third Example:

User call extension 500 directly.

Right away, call goes to GDM 500. User can leave a message.

(This is the only scenario that seem to work correctly)

Looking at this, to me it seems like perhaps the translation rule is not working, this is why example 2 gets that sound, I think. Example 1, I have no clue why the call is simply dropped.

Here is my current config:

voice hunt-group 5 parallel

final 500

list 512,513,514,516,517,518

timeout 300

pilot 501

voice translation-rule 4

rule 1 /^501$/ /500/

voice translation-profile GDM

translate redirect-called 4

dial-peer voice 100 voip

translation-profile outgoing GDM

destination-pattern 10.

session protocol sipv2

session target ipv4:10.10.100.208

dtmf-relay sip-notify

codec g711ulaw

no vad

ephone-dn  298

number 500

description GDM for group 501

call-forward all 100

call-forward busy 100

call-forward noan 100 timeout 300

!

Get these debugs to see the call flow:

debug voip ccapi inout

debug ccsip mess

The fact that dialing 500 directly works shows that the DN and CFB is working properly.  It sounds like something is up with the final not kicking in on the hunt group.  If the translation pattern was the issue, you'd hit CUE VM and it would say 'there is no mailbox associated with this extension.'

You may be hitting the max forwarding count?  Try configuring 'max-redirect 20' under telephony-service and trying again.

I had hope for this one. But it did not work. Same issue. Calling

501 directly and not letting anyone answer just goes to a fast busy sound.

Calling the AA and getting transfered to the hunt group, 501 and then

not letting anyone answer result in the call being dropped.

Is there any debug command I can see where the call gets dropped or how it's being routed?

Thanks

Is there any debug command I can see where the call gets dropped or how it's being routed?

See my previous post.  I mentioned the debugs you need to collect to troubleshoot this in that post.

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