Hello, I have been working on some changes to my CME setup, try to streamline somethings. But I have seems to now have broken voice mail. I have a DN that has a primary number (101) and a secondary number (1484XXXXXXX), the secondary number is a DID for the phone. I can not get a call to the DID to be answered by Unity. Looking at the logs, right before the SIP call is sent back to the provider I am seeing a 302 Moved Temporarily and I get the provider's voicemail that I use as a backup in case of any internet outages. I have the 101 as the extension number and the 1484 number as the E164 number. I don't think there are any issues with the communication between CME and CUE, if you call the main number I get the company AA, I can dial 101 from the greeting and it will move to that extension's mailbox without issue after the timeout. I can also call from another internal extension without issue. My guess is something is not right with the number config on the DN. Maybe there is better debugging I should be looking at. Any help would be greatly appreciated. Thank you everyon!
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Hello Everyone, I have had CME running for about a year now in my home/home office and for the most part, things are working okay. But I think there is room for improvement, and I am looking for your thoughts and suggestions. Here is the current setup: I have 4 SIP DIDs, 2120 (Business number), 2119 (Fax), 2112 (My personal DID), and 2113 (House mainline). 2120 goes to an AA running on CUE and 2119 to an analog port, I don't think either of these needs to be changed. Or I can't think of anything to do different there. If anything I guess I would like the business line to ring first and if no answer then to the AA. My private DID (2112) currently rings 2 phones in a hunt group, but this has been causing issues with the callerID information when the call goes to Unity. I get my extension information rather than the originating call information. I think I can fix this by only have the call go to a DN as a second button on my phone. I think I would still like the ability to answer from other phones, and I think that is what a group pickup is for but I am not sure. The main house line is where I think the most mess is, right now each IP phone has a default DN and that is assigned to button 1 of each phone, other than the 1 analog phone that is in the house. There is a hunt group that rings ext. 101-106 (105 being the analog phone) then to a AA to leave a message. So there are a few issues with this setup. 1) I can't add a BLF for the analog phone. 2) I am not sure if this is the best way to go at this. Knowing all of this is there anything you would do to streamline this setup? Please let me know if you need any additional information. Thank you
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I think I am so close to getting this working. The first dial-peer accepts the call from my provider. The invite comes to 1777XXXXXXX. I have attached a sip-profile to this dial peer:
voice class sip-profiles 2 request INVITE peer-header sip TO copy "sip:(.*)@" u01 request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
This dial-peer points to the IP address of the router as the session target
The second dial peer, which now has the incoming called number set to one of the DIDs and a translation profile that matches the called number to a DN. Looking at the logs I am seeing 484 Address incomplete errors.
If anyone could look at my debug logs, https://pastebin.com/UXU3gX1r and maybe provide a little guidance.
Thanks as always
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Ok so here is the problem, my SIP provider passes the DID number in the To: field of the header, so the issue is no matter what DID you dial CME see it as the user id number and not the DID, in turn, CME does not know how to route the call properly. The link that I posted, trying again, https://www.iptinfo.com/2014/03/04/callcentric-voip-with-cisco-unified-border-element-cube/ describes how to pull the data from the packet while passing from CUBE to CUCM. The issue that I have is that I have older gear, 1861 UC. So while I understand what is going on, I am trying to figure out how to do this while only having one device.
I think I can pull it all together if I can find how a call moves thru the router. For example, Call -> Voice Service/SIP config -> Dial-Peer -> Translation rules -> Call routed to DN/AA/Pilot or some other direction.
I think I understand all the parts and have the configure to pull the DID from the TO: field, just not sure where everything needs to go.
Please let me know if you have any other questions.
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The support forums, Reddit, and a little outside help has gotten me a long way in this project and I have learned a lot too. However, that being said the solutions we came up with was never really what I wanted (using VoIP gateways back to pots) so I have gone back to the drawing board as they say and I know with all of the really smart people out there a solution can be found.
This link will explain what is going on; https://www.iptinfo.com/2014/03/04/callcentric-voip-with-cisco-unified-border-element-cube/
What I am trying to put together is some config that takes CUBE and Call Manager and moves them together which is what I have (Cisco 1861 ISR). My production 1861 has CUE integrated my test 1861 does not. However, I don't believe that will cause an issue with testing with the calling side.
From the link above I was able to create all the translation profiles, I think where I am running into issues is the peers and if I need more than one. A Reddit user suggested creating a loopback interface. I think I get what he's thinking. Bring the call in, use the header manipulation rules, pass it back in thru the loopback then let CME process the call. Makes sense to me but not sure if that can be done. I look forward to your thoughts and suggestions.
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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:\email@example.com>"
voice class sip-copylist 2
That was then added to the pial-peer, but it doesn't seem to be doing anything. On to MWI.
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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.
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.
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.
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So after many years of want to learn, I picked up an 1861-UC and well all I can say is WOW. Had now ideas how different this could be from my ASA. And I will admit I have gotten pretty good with the ASA over the last few years.
I will admit I have always like Cisco IP phones and I have a few now running on SIP with Asterisk. But I have always heard that SIP and Cisco phones are not the best combination. So I was looking around and found an 1861-UC for a good price and thought why not let's learn something new. Now I understand these are end-of-life and have been for a long time but that's ok this is just for home and home business use.
So let me explain a little of what I have and maybe I can get some recommendations on how to best set this up or at least a framework that I can work with. I have been able to get the Cisco phones to get an IP address from the router, download their configs and call each other. Custom screen backgrounds, ringtones, and MOH are for another time. And speaking of time that is off too.
Currently, I have 3 DIDs from call-centric; personal (home), business (home business) and fax. Lets start with the fax number as it is the simplest, a call comes in from that DID route it to the FXS port the fax is plugged into. A personal call comes it, rings phones that use the personal line, say ephone 1, 2, 3, and 4. These phones would have a line appearance and then that line is selected the outbound caller id is the personal number. Same would be for business but only phones 1 and 3 ring and they have a line appearance for it. When that button is used to place a call the business number is shown for outbound caller id.
There is also the issue of getting the SIP trunk up and running. I could use an ATA an use the FXO ports, but I would like to get this working without that. Less hardware to maintain I guess. Thank you for your time and I look forward to getting this working.
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