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Walkthrough Wednesdays

Cisco 1861 CME and CUE

Hello Everyone,


First off I understand that I have outdated equipment, but at this time and as a learning tool this is all that I wanted to spend. I will be honest I have always had a liking of Cisco telephony and wanted to get my feet wet. Although I think I have jumped in head first. Up until that last few weeks, my VoIP experience has been with Asterisk and FreePBX and while they do what I need them to do I have never been extremely happy with it. Never really felt like a complete finished product. So I had the opportunity to pick up a cheap Cisco 1861 ISR and some phones, and I have been banging my head off the wall ever since. Sorry, this will be a long post, but there is a lot of base information to get out.


The Plan

I run a small home business, that someday I would like to expand. Currently, I have my family helping out with calls so I have some phones at the desks of family members (not that this is too important). I also have a friend off-site that will be helping soon, so I will be looking at adding a remote user at some point. I have currently 4 DIDs from my SIP provider (CallCentric). The first 1XXXXXX1212 that is my personal line, I am the only one to have a direct DID which could change. The second 1XXXXXX2120 is the business mainline, and the third 1XXXXXX2119 is the fax line DID. The last number is 1XXXXXX9187 which will be set up as an email to phone service that will ring that 'Emergency' number if one of the sites I host has an issue or goes off-line.


Here is what my flow looks like:

  • 1212 DID - Ring my extension (101), busy or no answer transfer to my mailbox
  • 2120 DID - Auto Attendant
    • 1 Company Directory
      • Dial by name
    • 2 Ring phone with Mainline DN button assigned, if busy/no answer transfer to the general mailbox
    • 9 (After hours) Transfer to the general mailbox
  • 2119 DID - Ring FXS port 0/0/0
  • 9187 DID - Ring phones with Emergency DN button assigned

What is working - Start with the good stuff

  • Call 2112, 2120, and 9187, ringing the proper places
  • Calls internally are working and transferring to the mailbox on busy/no answer
  • I am pretty sure the AA script is working as intended, I will need to test that some more to be sure
  • Clock is showing correct time and date!!!

What's NOT working

  • Outbound Caller ID - no matter what DN I select I get the 2112 DID
  • Incoming call not transferred to my mailbox - of you call the 2112 DID, my phone rings as planned, however, the call is never transferred to my mailbox. If I call my extension internally call is transferred to my mailbox
  • Incoming call not terminated - if you call a DID and hang up the call continues to ring, sometimes 2 or 3 times, sometimes indefinitely until you pick up the phone
  • '9' for secondary dial tone - I think this may be related to the outbound caller ID DID, however, when you dial '9' to get the secondary dial tone, then dial 1NXXNXXXXXX I am getting an 'Unknown Number' error
  • FXS port - nothing happens there, nothing on inbound, outbound soon as I go off-hook busy signal
  • Mailbox notification - the phones do not show voicemail notification on new message, I would also like some information on email notification for voicemail.

Thank you

If you have made it this far in my very long post I want to thank you. I do not expect anyone to just help me fix all of this so I am willing to pay for services and help. I, unfortunately, can not afford normal rates so I completely understand if no one wants to take this on. There is still a lot of work to do in making CME what I would like it to be. I have attached a current running config, and I look forward to anyone's help, suggestions, and guidance. Please let me know if there is anything else anyone needs to know. I really am getting to the point of packing this all up. I guess this is why noobs shouldn't play with the big boys (and girls).


Again, thank you for reading.


VIP Collaborator

Easy stuff first:


Outbound Caller ID - no matter what DN I select I get the 2112 DID


Check with your service provider to see if they are setting callerID. It is common for a service provider to set the CLID to the "main line" DID number for a bank of DIDs like yours. You may be able to ask that they not do this for the one DID or for all DIDs.


Mailbox notification - the phones do not show voicemail notification on new message, I would also like some information on email notification for voicemail.

I assume you are using Unity Express for voicemail. In your config I do not see ephone-dn's configured for MWI on and MWI off. The 999 and 998 in the example below are just examples. These mwi on/off numbers must match what is configured in CUE. And the number of dots after the 999/998 must match the number of digits in your extensions, so 3 dots.

ephone-dn xx
 number 999...
 mwi on

ephone-dn xx
 number 998...
 mwi off

ccn application ciscomwiapplication aa
 parameter "strMWI_ON_DN" "999"
 parameter "strMWI_OFF_DN" "998"
 end application

ccn trigger sip phonenumber 801
 application "voicemail"
 end trigger

Also, while it looks to me like you are using CUE for voicemail, I see this under sip-ua:
 mwi-server expires 3600 port 5060 transport udp

If you are using CUE for voicemail I think it's supposed to look like this:
 mwi-server ipv4: transport udp port 5060
 number 2010


I'm still looking at the rest....




Thank you for the reply and I will begin to look into the MWI information. I did want to touch on the Caller ID so my provider uses the Remote-Party-ID for the outbound caller ID DID, I did find a post online that recommended this:


voice class sip-profiles 3
 request INVITE peer-header sip FROM copy "sip:(.*)@" u02 
 request INVITE sip-header Remote-Party-ID modify "<sip:(.*)@(.*)>" "<sip:\u02@>" 

voice class sip-copylist 2
 sip-header FROM

That was then added to the pial-peer, but it doesn't seem to be doing anything. On to MWI. 



It really depends what the IOS is sending in the RPID. That sip profile copies the info in the FROM header, stores it in a "u02" variable and then uses this to modify the RPID header potentially for outbound SIP INVITE's.
Your CME SIP call logs need to be analysed to see what is going on.

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"
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