I'm looking for some help configuring a voip.ms incoming trunk line with CUCM 7.0.1 . Outgoing is working fine. Screenshots of the config are attached. Voip.ms allows static authentication, and I see the packets coming into the server which is not behind a nat, but I'm getting a busy tone.
I think I configured everything I need, hopefully properly, so any insight would be greatly appreciated. I tried to take screenshots of everything I did, so its likely that if its not attached I didn't config.
One thing I noticed that I can't figure out is that when you are calling extension to extension is that the phone you are calling the user's name appears on the calling phone, but the extension name shows on the destination phone. The two phones are set up the same aside from the ext #s.
Thanks in advance.
No, it has nothing to do with dial plan. Your SIP trunk is using "search space" as the CSS, your phone is in internal partition. You need to ensure that whatever the dialed number is when it hits the GW i.e. 102 that this DN is in partition assigned to the Search space CSS. Can you provide the screen shot of this CSS configuration?
Also, are you expecting to see only 3 digits from provider, that is not very typically, whatever they send you you need to match exactly either by the DN on the phone or running it through translation pattern or setting the Significant Digits field on the SIP trunk to i.e. 3 which will only preserve the last 3 digits.
Thanks for the reply. I have "local partition" as the selected partition for search space. See screen shots.
I'm not expecting 3 digits, but 10. I'm seeing on the SIP trunk significant digits is set to all. Is the problem in the DN?
Yes, "All" means CUCM will called all digits presented in your case 10 and using the CSS will attempt to route to next hop. If you do not have any DNs or patterns matching this the call will fail.
If your DNs are always the last 3 digits of the DID then you can set the significant digits field to 3, otherwise you can create a translation pattern that matches the 10 digits and transforms it to the DN, don't forget to put the TP in proper partition and assigned CSS that has access to the DN.
HTH, please rate all useful posts!
Thanks again for the reply.
I think you're giving me too much benefit of the doubt - you lost me. Without translation patterns, should I just create the first directory # to be the 10 digit outside line #, and on the directory number configuration page make sure its in local partition/search space as configured?
No need to change DNs.
Without Translation Patterns, on the SIP trunk configuration page change the Inbound Calls --> Significant Digits field from "All" to 3, it's a drop down box that allows you to set the number of digits you want to route based on.
If you want to use Translation Patterns go under Call Routing --> Translation Patterns and add new one where the pattern will be whatever you are getting from telco, let's say you get 2125558XXX where XXX are your defined DNs, i.e. 102, then the pattern would be 2155558XXX in local partition, where the transformation is set to XXX, basically it will take all 10 digits matching 2155558XXX and transform it to preserve the last 3 digits, then this TP need to have a CSS assigned to it that can reach the 102, etc DN.
Thanks for the reply and the patience. I'm learning by doing here with call manager... and the documentation only gets me so far.
The line doesn't end with 102, so what I did was changed the DN to the last 3 digits , 960. Then I added that DN to local partition, etc. I then added the translation pattern you described (shown in attached screenshot). Do I leave significant digits to all or change it to 10? I'm still getting a busy signal after a reset.
If you chaged the DN to be the last 3 digits of the DID, and changed the sig Digits to 3 there is no need for the TP.
Let's start with basics:
1. define the SIP trunk with sig digits = 3
2. Set the CSS on the Sip trunk to "None"
3. Reset the SIP trunk
4. Define the DN to match the last 3 digtis of the dialed number, i.e. 960 assign it to "none" parition.
Now call the number and let's see if that works, if it does not we will need to look at your CM SDI traces.
No luck. Also, the phones won't ring each other with the extensions anymore. I'm tempted to nuke this and start from scratch - thoughts? Do you know if there are a good set of instructions anywhere to follow in setting this up? I haven't been able to find any.
Can you do the ff:
on your voice gateway
1. debug ccsim messages
2. debug voip ccapi inout
do a test call...
send the output of the debug here with the called and calling number..
Thanks for the replies. I feel like I had so many config errors it would be easier to start from scratch... and do it correctly. I have a clean install completed and ready to go. I've spent hours looking for a good tutorial on how to get this done and even thumbed through a CCNA voice textbook which was rather unhelpful. Can anyone either point me in the direction of a good tutorial or spare a few minutes and help me out? I appreciate the responses, and all the help.
Back to square one. I reinstalled fresh and have outbound calls and internal calling working agin. This is what I've done so far. I'm hoping someone can give me specific instructions on how to get the incoming calling working.
1- Configure SIP Trunk security profile
2 - add sip trunk (4 significant digits), route partition
3 - route pattern [2-9]xx[2-9]xxxxxx, route partition
4 - added phones. First phone's DN is 8960 (last 4 of DID), second phone is 8961 (last 4 of 2nd DID). The phones can now call each other and call outbound.
If I call in, I get a busy signal. I see the packet from voip.ms come in when I place the call, and it hit the CUCM, but it ends there.
All help is greatly appreciated.
You may be very near success if you have outbound working on the SIP trunk. Good job. I hope the following may help you fix the inbound flow. There are good books from Cisco Press on the subject (e.g. SIP Trunking, Cisco Voice Gateways & Gatekeepers, etc.), but sometimes learning by doing is the most fun --as long as it is not done at a customer site
If inbound isn't working, I like to examine the incoming URI found in the SIP INVITE message.
Try this if possible on the voice gateway which terminates the SIP trunk on your side:
1. Activate text capture on your terminal emulator program or increase the scrollback buffer to at least 2000 lines. You're gonna need it.
2. Access IOS enable mode of the gateway
3. term mon
4. debug ccsip messages
5. Place your inbound test call
6. undebug all <-- turn off the debugs
There are slicker ways to enable logging on the voice gateway than that, but this should work.
Now see if you can find the invite in the log you captured. It might look like this:
r3845#debug ccsip messages
SIP Call messages tracing is enabled
000069: May 8 01:35:47.215: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
INVITE sip:email@example.com:5060 SIP/2.0
See the invite message? The digits preceeding the @ sign are like DNIS digits being sent to you (e.g. 4198675309). That number must exactly match something in your CUCM config, or else you must use one of several mechanisms to translate those digits to a number that does match a number in your CUCM. One or the other -- it must match something.
I am assuming you rebuilt the SIP trunk, but I am unclear how the Significant Digits are currently configured. That's a VIP --a very important parameter for inbound to work. You can always select Help>This Page in CUCM Administration for a quick read how this works.
Basically, figure out what the other end is sending you in the URI. Then, a crude easy fix is to adjust the significant digits so that the "least significant" digits in the URI your provider sent "land" on one of your directory numbers. Else, you may need to research & employ either IOS voice translation profiles or CUCM translation patterns.
I am taking a giant guess that this is what you need to do, given that the SIP trunk works outbound and you see a packet arrive inbound. Your mileage may vary. Good luck. HTH TZ
Thanks for the reply TZ. It certainly is much more fun learning this way than buying those completed lab kits on ebay.
Since I changed the DN's to match the last 4 digits of the phone #s, I have significant digits on the SIP trunk set to 4. With a little help, I managed to get some logs and it looks like a NAT problem now (or at least first). I've got a 1-1 nat set up for the CUCM server, but it apparently doesn't like the external ip coming in rather than its IP...
05/07/2012 20:49:57.500 CCM|Digit Analysis: Host Address=96.250.50.xxx DOES NOT MATCH any address for this cluster.
Added a SIP-ALG proxy on the router and its all working now, sort of. I'm getting each phone to ring with the proper DID. the outbound voice works, but I cannot hear the incoming voice.