CUCM 8.x Setup

1) You need a separate system called Unity Connection. From the same installation disc from which you built the CUCM server, you can choose to install Unity as a standalone server, this way both server can be part of clusters, or, you may choose to install UCM and UC on the same machine and have a co resident install that is called a CUCM Business Edition. No cluster possibility at all.

2) What do you mean, like getting a second dial tone when calling outside numbers?

1) Ok let me see if I have this... so I will have to install another VM with this server?

2) Yes, which I assume we need to get a SIP trunk.

1) Yes.
2) Only 2 type of patterns that will provide secondary dial tone are Call Rouring > Translation Patterns and Route Patterns. There is a checkbox called provide secondary tone you will want to check.

2) Actually, I am just trying to setup to call out. I have a SIP provider, but I want to start receiving and sending calls. I am wondering how to configure this so we can start calling.

That's  an item that deserves its own thread/question.... Nevertheless, first thing you may want to search is documentation from your SIP provider regarding CUCM Configuration. If none, start off by creating a new SIP trunk in CUCM and point it to your provider's IP address they should have shared with you. Create a Route Pattern and point to it to the SIP trunk.

Can you explain how to get incoming calls to work? I am able to make outgoing calls through my provider, voip.ms

It would be great to get a step by step.

I am new to this as coming from the CCNA world....

I know our provider uses IP authentication...so I am not going through a gateway or anything else for that matter than a Cisco E3000 to Comcast.

I raised the point value btw.

To handle incoming calls is just a matter of determining which numbers will your ITSP be calling. For example, lets say they provided you with the following number of 2223334455, so people will contact you at that number right? Most likely, on incoming calls you will receive the full 10 digit string as the called party.  Go ahead and configure an IP Phone with directory number 2223334455, it should ring on right on when you call it from your cell phone.

What if my phones have extensions running off one DID? I assume that is where we use the features of the CUCM?

I have about 4 phones to run off one DID. I would like to have someone call, and have it hunt through these.

My phones are ordered 1000-1004

Yeah thats possible too, and that is where we are heading, but if you are having troubles with incoming calls, the scenario I suggested above will provide the simplest possible call flow to analyze what is wrong.

I configured directory number as XXX-XXX-XXXX.....that is all and when I called form cell it did not ring.

Also put my 10.0.0.25, or my CUCM, in the DMZ of the router.

Just a slow busy.

Ok, please go ahead and open an ssh connection to the UCM server, then follow this instructions:

https://supportforums.cisco.com/docs/DOC-11599

Once you reproduce the problem by calling in while capturing packets, please download the resulting file and upload it to cloudshark.org. I will be able to analyze it provided the IP addresses of the UCM server, ITSP, phone, and any NAT translation in between.

Let me know if this works.

http://cloudshark.org/captures/1813266988d6

I made 1 or 2 calls and cut it, as it was producing rather large files for upload.

The ip address of the UCM is 10.0.0.25

Unity is 10.0.0.24

SBS 2011 is 10.0.0.2 with DNS

My PC which also hosts the 2 VM's of UCM and Unity is 10.0.0.10

and router is 10.0.0.1

I just made outgoing calls from cell to our DID which is 734-794-7177. Again, outgoing works just fine.

See if this help as well. I filtered my 10.0.0.10 address. To show traffic from router to CUCM.

http://cloudshark.org/captures/63f3b899581c

I made a call first to my CUCM from cell. (Slow busy)

Next I made a call from my IP comm soft phone to my cell. (Suceeded)

Neither capture contains the calls we are after. The second one contains an outgoing call only:

113		51.531347	10.0.0.25	64.120.22.242	SIP/SDP	1152	Request: INVITE sip:15173914946@64.120.22.242:5060, with session description
114		51.567424	64.120.22.242	10.0.0.25	SIP	509	Status: 100 Trying
117		56.300534	64.120.22.242	10.0.0.25	SIP/SDP	807	Status: 183 Session Progress, with session description
931		63.559973	10.0.0.25	64.120.22.242	SIP	408	Request: CANCEL sip:15173914946@64.120.22.242:5060
935		63.602823	64.120.22.242	10.0.0.25	SIP	494	Status: 487 Request Terminated
936		63.603483	10.0.0.25	64.120.22.242	SIP	441	Request: ACK sip:15173914946@64.120.22.242:5060
937		63.604000	64.120.22.242	10.0.0.25	SIP	478	Status: 200 OK

Select allOpen in new window

Please do not capture outgoing calls. Start the sniffer first, then call in from your cell phone, wait for it to ring busy, then stop the capture.

Thanks,

Ok so I added the SIP URI in vopi.ms settings and it seems to now have some traffic, but rejected.

http://cloudshark.org/captures/1aa63d91a413

Looking at the message data on the packets...

Message Header
Via: SIP/2.0/UDP 64.120.22.242:5060;branch=z9hG4bK4368394a;rport
From: "5173914946" <sip:5173914946@64.120.22.242>;tag=as2549e769
To: <sip:17347947177@68.62.101.196>;tag=15~734cdcf4-bdd6-4693-96e8-65d7427b8f31-33073141
Date: Fri, 25 May 2012 02:30:44 GMT
Call-ID: 4f4d679e1d87601c279bf7e87113124d@64.120.22.242
CSeq: 102 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0

.It may seem that I could be misconfigured on my side. I dont have any translations or any settings specifically for incoming calls. So Maybe they dont know how to route?

Q.850 Cause Code 1
Q.850 Release Cause Description
Unallocated (unassigned) number

Typical scenarios include:

•The number is not in the routing table, or it has no path across the ISDN network.

Indicates that the destination requested by the calling user cannot be reached because the number is unassigned.

I have one of my Cisco IP communicator soft phones configured with the directory number of 7347947177

Do I need to add any settings for incoming?

That is not the correct number. They are looking for 17347947177. Add a 1 to the IP Communicator and try again to see if it works this time. Reset the device before doing it. Make sure the line is not associated to a route partition at this time.

I tried adding it....geez, and it still is slow busy. What else can we try?

Just to be sure, I dont need anything in the Route/Hunt to be configured at all?

The only thing I have configured under Call Routing menu nd Route/Hunt is

A) Line Group for Voicemail

B) Route pattern for outgoing calls

Remember I told you at this point we dont want to make it more messy than it is. Just a phone ringing for now. Please hand me a set of CallManager traces this time:

trace config: http://tools.cisco.com/squish/f3537
trace collection: http://supportforums.cisco.com/docs/DOC-11588
Go to System > Service Parameters > choose the UCM IP > Choose the CallManager service > go to the Digit Analysis Complexity  parameter and set it to the other non-default value there is.
Capture an incoming call and attach the traces here please.


Thanks,

When I go to remote browse, which options do I check for trace?

Cisco Callmanager, once you get the list of files, please only download SDI traces, not SDL

Got this error trying to upload zip....The extension of one or more files in the archive is not in the list of allowed extensions: cucm8/2012-05-25_18-49-12/cm/trace/ccm/calllogs/calllogs_00000001.txt.gz

Anyhoo...uploaded to file share...here is the link:

http://www.filefactory.com/file/c5a27e6/n/cucm8_zip

I called around 2 times to my DID

Hey, I'm seeing some errors like this:

17:56:10.295 |!!ERROR!! -ConnectionManager- wait_AuDisconnectRequest

However, the trace filters don seem to be set up correctly. Are you sure you have them set to detailed level as explained here: http://tools.cisco.com/squish/f3537

Also, please include all checkboxes related to SIP. Please try again, Im sure will be able to move forward once we get the traces.

Oh, and this time, please just change the extension of the file to PNG to be able to attach in here. No external servers required. And please only include the SDI trace files.

From UCM I switched the Digit Analysis Complexity  from StandardAnalysis to TranslationandAlternatePAtternAnalysis. Those two were the only options.

I Made sure to uncheck trace on for SDL....I checked everything on SDI.

Verified everything. I re-ran two calls in to my DID.

Attaching results. (If you want to remote in to my system to look, you can...I can setup a remote connectivity...just let me know. You may find something I am not telling you.)
ccm00000001.txt
calllogs-00000001.txt

Hopefully you can help me fix this! lol

Ok so supposedly you can get CUCM to work with IP authentication which I do have on voip.ms.

What do I need to do from your view point?

Do I have something Ill configured?

My understanding is that I do not need a CUBE for IP auth

Yes, but IP auth has nothing to do here. The device you have connecting your internal network to the Internet is breaking the communication. You need an Application Layer Gateway (ALG), a CUBE, a STUN or ICE system that will help you fix the NAT. UCM doesn't suport STUN or ICE however. You need a more powerful device that will inspect packets at layer 5,4 and 3 instead of only 3 and 4. Your configuration is ok it seems.

My understanding is that I do not need a CUBE for IP auth

Its not IP auth we are talking about, its network address translation what we are trying to fix.

REgards,

Roger. What would you recommend, and can anything be run via software not hardware like on a VM?

Would you think assigning a machine with a separate external static IP would work as well?

We are a small office with 10-15 phones and need to go with the cheaper route.

You may want to read about www.opensips.org. If you configure it simply as a SIP proxy with very basic settings, you can use it to re-write the SIP headers containing public IPs and then pass the traffic to UCM.

Wow, ok so being a on linux expert. it seems a little daunting.

Could you recommend something a little more straight forward? Or maybe an idea of how to configure?

http://www.voip-info.org/wiki/view/Asterisk+sip+nat

http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

http://www.linuxjournal.com/article/9399

Hey Im sorry, I threw out a few recommendations for Asterisk, I got it mixed up. Let me know if the recommendations so far have put you on the right track.

Regards,

Any other recommendations?

http://www.cisco.com/cisco/web/solutions/small_business/products/voice_conferencing/uc_300_series/index.html