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Avatar of abhijitm00
abhijitm00🇺🇸

Adding new DIDs to PRI
Hi Experts,

We have Cisco Call Manager 4.2 running. We currently have 3 PRIs, we currently ordered 50 new DIDs from a different exchange (315-761-9450 through 9499) and I need to set it up. The telco says that the numbers have been applied to our PRI, when I dial in with a number it gives me "We are unable to complete your call at this time, please try again later" I believe this has something to do with translation pattern. I have put in 215761XXXX but that is not working.

What should I be looking at to troubleshoot this? We recieve last 4 digits from telco. Thanks

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Avatar of Deuce08Deuce08🇺🇸

Sounds to me that you just need to configure a DN on the Call Manager side, then when you dial it externally it will pass across the PRI and deliver the 4 digits to your system.

Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Thanks for getting back. I have set up a number on my phone but on dialing from outside I still get that error message above. Is there somewhere I can look to see if it is even recieving the digits?

Avatar of Deuce08Deuce08🇺🇸

Sure is, but you have to run the commands on the router hosting the PRI.  Before we get there, got a couple questions though.

Is it a Cisco router?  If yes, what series?
Did the telco just ADD these DIDs to existing PRIs or did you get new PRIs as well?

If it's a Cisco router, I'm good and can give  you some debug commands to run.

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Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Thanks Deuce. It is a Cisco 2821 Router. These are existing PRIs, no new PRIs were added.

Avatar of Deuce08Deuce08🇺🇸

Below are a the 2 debug commands for call presentation, after entering the debug commands, have the terminal monitor command ready and dial one of the DID numbers.  If you have as many calls going through as I think, you'll want to wait to monitor it to just before the call is placed.

After you place the call, wait a few seconds and type terminal no monitor (you will not be able to see the command as you type it) and copy/paste your debug results to notepad to search for the DID number.

debug isdn q931debug isdn q921

You should see something similar to below....
 
Oct 30 16:51:49.592: ISDN Se0/0/0:23 Q931: SETUP pd = 8  callref = 0x4CE2
        Bearer Capability i = 0x9090A2
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98394
                Exclusive, Channel 20
        Facility i = 0x9F8B0100A10F02019706072A8648CE1500040A0100
                Protocol Profile =  Networking Extensions
                0xA10F02019706072A8648CE1500040A0100
                Component = Invoke component
                        Invoke Id = 65431
                        Operation = InformationFollowing (calling_name)
                                Name information in subsequent FACILITY message
        Progress Ind i = 0x8483 - Origination address is non-ISDN
********Calling Party Number i = 0x2181, '7345551212'
                Plan:ISDN, Type:National
********Called Party Number i = 0xC1, '1665' <---------THIS IS DID*****
                Plan:ISDN, Type:Subscriber(local)
Oct 30 16:51:50.168: ISDN Se0/0/0:23 Q931: FACILITY pd = 8  callref = 0x4CE2
        Facility i = 0x9F8B0100A11802020099020100800F4B4F4348204A45464652455920414C
                Protocol Profile =  Networking Extensions
                0xA11802020099020100800F4B4F4348204A45464652455920414C
                Component = Invoke component
                        Invoke Id = 153
                        Operation = CallingName
                                Name Presentation Allowed Extended
                                Name =  CALLER ID NAME***************
Oct 30 16:51:57.255: %ISDN-6-CONNECT: Interface Serial0/0/0:19 is now connected to 7345551212 N/A

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Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Very useful. Thanks. How should I go about it if I can't see the 4 digits showing up. Would I look into the translation pattern or something else? Thanks

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Avatar of Deuce08Deuce08🇺🇸

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Avatar of benhansonbenhanson

There is actually quite a bit involved.  You probably have a covering translation pattern that gets it from your gateway to an internal calling search space.  Depending on how specific the covering translation pattern is, you may need to add another translation pattern to throw the call into your internal calling search space.

Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Hi Deuce,

It tried out the commands you mentioned and this is the information I recieved from the debug command:

ISDN Se0/1/1:23 Q931: RX <- SETUP pd = 8  callref
= 0x0384
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Progress Ind i = 0x8283 - Origination address is non-ISDN
        Calling Party Number i = 0x2181, '2155671063' - MY CELL NUMBER
                Plan:ISDN, Type:National
        Called Party Number i = 0x80, '9452' ------------- DID NUMBER
                Plan:Unknown, Type:Unknown
1165412: Nov  2 20:06:07.570: ISDN Se0/1/1:23 Q931: TX -> SETUP_ACK pd = 8  call
ref = 0x8384
        Channel ID i = 0xA98381
                Exclusive, Channel 1

When I dial out from the 9452 phone then I can establish a call but when dialing to this number I get the voice message as above

Avatar of abhijitm00abhijitm00🇺🇸

ASKER

There was some more at the end:

Cause i = 0x82E1 - Message type not implemented
        Call State i = 0x06
1165569: Nov  2 20:18:36.810: ISDN Se0/1/1:23 Q931: TX -> DISCONNECT pd = 8  cal
lref = 0x8388
        Cause i = 0x80E5 - Message not compatible with call state
1165570: Nov  2 20:18:37.614: ISDN Se0/1/1:23 Q931: RX <- RELEASE pd = 8  callre
f = 0x0388
1165571: Nov  2 20:18:37.618: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd = 8  c
allref = 0x8388

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Avatar of Deuce08Deuce08🇺🇸

Can you provide the router config....you may be missing the dial peer association in the router to pass the call in.

Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Please find attached. Thanks,
shrunrouter.txt

Avatar of Deuce08Deuce08🇺🇸

One thing I see missing is in fact the incoming dial-peer.  I am not sure how detrimental this is not having it, but it's something that wouldn't hurt to throw in.  Other than that, I only had a few seconds to look it over, I don't see anything missing.  Try it and let me know, be sure to get the period (.) on line 3.
dial-peer voice 1 pots
 description **** Incoming from PSTN ****
 incoming called-number .
 direct-inward-dial
 port 0/1/1:23

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Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Ok I will try that, its just that we have a separate list of DIDs (315-587-8800 to 8950) and the dial peer information is not on the gateway. Do you know what the following would mean?

- Overlap Receiving Flag for PRI* on the CallManager Service Parameters Configuration page.
-  IE Delivery in the gateway configuration page in the Cisco CallManager. ?
- Check Send Extra Leading Character In DisplayIE.

Thanks

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Avatar of abhijitm00abhijitm00🇺🇸

ASKER

Ok its working now. I needed to add the partition to the correct Calling search space as afrtadmin mentioned. Now I am able to recieve calls on those DIDs. I have split the points between the two of you. Thanks.
Voice Over IP

Voice Over IP

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Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).