VOL1 3.7 Globalization

Globalized Call Routing Step 8: Device > Gateway. Select the MGCP Gateway for R1. Choose the Called and Calling party transformation CSS's

It shows CSS_HQ_CALLED and CSS_HQ_CALLING as the two used.  We were not instructed to create these CSS's a few steps earlier.  It would have to be which two of the following?

CSS_HQ_CALLED_IN

CSS_HQ_CALLED_OUT

CSS_HQ_CALLING_IN

CSS_HQ_CALLING_OUT

 

 

Comments

  • Hmm, this task (3.7) feels very "disconnected" from all the previous tasks.  I had to undo everything I did in 3.1 - 3.6 as this one uses all new PT and CSS.  I'm going to go through it again but at this point nothing is working.

  • Mark,

    I believe they would have been CSS_HQ_CALLED_OUT and CSS_HQ_CALLING_OUT, though I don't have the book in front of me at them moment. 

    Mark Snow
    CCIE #14073 (Voice, Security)
    Instructor
    Internetwork Expert

    On Dec 14, 2009, at 20:05, MarkH <[email protected]> wrote:

    CSS_HQ_CALLED_OUT
  • It's Called_out per Josh's reply to someone a week or so ago.

     

    http://ieoc.com/forums/p/9294/82526.aspx#82526

  • and yeah, i actually just did globalization in another session...everything is different.

  • A couple of things to note.  Josh's verification in the other post is for Step 7.  I went with Mark's suggestion for Step 8 and assigned CSS_HQ_CALLED and CSS_HQ_CALLING to Device > Gateway > Transformation CSS.  When I place a call from HqPh1 to PSTN 3122012222 I get 'Your call cannot be completed as dialed.'  

    Below are ISDN debugs on both r1-hq and PSTN routers for a test call that fails.

     

     


    PSTN# debug isdn q931

    Dec 15 05:41:57.136: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x0007 

    Bearer Capability i = 0x8090A2 

    Standard = CCITT 

    Transfer Capability = Speech  

    Transfer Mode = Circuit 

    Transfer Rate = 64 kbit/s 

    Channel ID i = 0xA98383 

    Exclusive, Channel 3 

    Display i = '1001' 

    Calling Party Number i = 0x0081, '1001' 

    Plan:Unknown, Type:Unknown 

    Called Party Number i = 0xA0, '+17755011111' 

    Plan:Unknown, Type:National

    Dec 15 05:41:57.152: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0x8007 

    Channel ID i = 0xA98383 

    Exclusive, Channel 3

    Dec 15 05:41:57.156: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x8007 

    Cause i = 0x8281 - Unallocated/unassigned number

    Dec 15 05:41:57.192: ISDN Se0/2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x0007

    Dec 15 05:41:57.196: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8007







    r1-hq# debug isdn q931

    Dec 15 05:43:31.772: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x0009 

    Bearer Capability i = 0x8090A2 

    Standard = CCITT 

    Transfer Capability = Speech  

    Transfer Mode = Circuit 

    Transfer Rate = 64 kbit/s 

    Channel ID i = 0xA98383 

    Exclusive, Channel 3 

    Display i = '1001' 

    Calling Party Number i = 0x0081, '1001' 

    Plan:Unknown, Type:Unknown 

    Called Party Number i = 0xA0, '+17755011111' 

    Plan:Unknown, Type:National

    Dec 15 05:43:31.788: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0x8009 

    Channel ID i = 0xA98383 

    Exclusive, Channel 3

    Dec 15 05:43:31.792: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x8009 

    Cause i = 0x8281 - Unallocated/unassigned number

    Dec 15 05:43:31.828: ISDN Se0/2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x0009

    Dec 15 05:43:31.832: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8009




     

    Two things wrong. One is the calling number is 1001 instead of 7752011001.  Second is the the calls fails because the PSTN router says Unallocated/unassigned number. Is the + supposed to make it all the way through to the PSTN router? I assume yes.  I'm using the router configs from the Member's area (dated August 11).  Hopefully this is still the correct set of configs.

     

    EDIT: VoIP to PSTN is still not functioning but PSTN to VoIP is working as expected. For the very last step I tried pattern +1[2-9]XX.[2-9]XXXXXX just to make sure my head is screwed on straight.  Thankfully it worked.  Now, that pesky VoIP to PSTN is really gettin' to me.  Will resume in the morning. [|-)]

  • My apologies about the reference to the wrong step.

     

    As for the calling number, somewhere the "Use External Phone Number Mask" box is not checked...or, is it set on extension 1001 at all?

     

     

  • Mark,

    Sorry about that if it was the wrong CSS, I didn't have the book in front of me at the moment. 

    The important thing for you studies, is to trace whatever CSS you assign to outbound Called Party Transformation CSS on the Gateway, back to a Partition that contains a Called Party Transformation Pattern and that pattern should do the following:

    Match this number: +17755011111 (which is a local number I believe, so really you don't need to match everything, just enough to match all local called numbers)

    So a pattern like this should do the trick:
    +1775.xxxxxxx
    DD=PreDot
    Type=National



    Remember, the idea isn't just to follow the solutions in the book (not implying that that's all your doing), but rather to get it working to begin with and then use the solutions as a quick reference.
    I will of course take a look at that part of the sols guide and see where the disconnect is.


    Cheers,


    Mark Snow, CCIE #14073
    Instructor
    Internetwork Expert, Inc.
    mailto:[email protected]
    http://blog.ine.com
    Toll Free: 877-224-8987
    Outside US: 775-826-4344



    On Dec 15, 2009, at 12:40 AM, MarkH wrote:

    A couple of things to note.  Josh's verification in the other post is for Step 7.  I went with Mark's suggestion for Step 8 and assigned CSS_HQ_CALLED and CSS_HQ_CALLING to Device > Gateway > Transformation CSS.  When I place a call from HqPh1 to PSTN 3122012222 I get 'Your call cannot be completed as dialed.'  

    Below are ISDN debugs on both r1-hq and PSTN routers for a test call that fails.

     

    PSTN# debug isdn q931
    Dec 15 05:41:57.136: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x0007 
    Bearer Capability i = 0x8090A2 
    Standard = CCITT 
    Transfer Capability = Speech  
    Transfer Mode = Circuit 
    Transfer Rate = 64 kbit/s 
    Channel ID i = 0xA98383 
    Exclusive, Channel 3 
    Display i = '1001' 
    Calling Party Number i = 0x0081, '1001' 
    Plan:Unknown, Type:Unknown 
    Called Party Number i = 0xA0, '+17755011111' 
    Plan:Unknown, Type:National
    Dec 15 05:41:57.152: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0x8007 
    Channel ID i = 0xA98383 
    Exclusive, Channel 3
    Dec 15 05:41:57.156: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x8007 
    Cause i = 0x8281 - Unallocated/unassigned number
    Dec 15 05:41:57.192: ISDN Se0/2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x0007
    Dec 15 05:41:57.196: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8007
    r1-hq# debug isdn q931
    Dec 15 05:43:31.772: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x0009 
    Bearer Capability i = 0x8090A2 
    Standard = CCITT 
    Transfer Capability = Speech  
    Transfer Mode = Circuit 
    Transfer Rate = 64 kbit/s 
    Channel ID i = 0xA98383 
    Exclusive, Channel 3 
    Display i = '1001' 
    Calling Party Number i = 0x0081, '1001' 
    Plan:Unknown, Type:Unknown 
    Called Party Number i = 0xA0, '+17755011111' 
    Plan:Unknown, Type:National
    Dec 15 05:43:31.788: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0x8009 
    Channel ID i = 0xA98383 
    Exclusive, Channel 3
    Dec 15 05:43:31.792: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x8009 
    Cause i = 0x8281 - Unallocated/unassigned number
    Dec 15 05:43:31.828: ISDN Se0/2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x0009
    Dec 15 05:43:31.832: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8009
    Two things wrong. One is the calling number is 1001 instead of 7752011011.  Second is the the calls fails because the PSTN router says Unallocated/unassigned number. Is the + supposed to make it all the way through to the PSTN router? I assume yes.  I'm using the default PSTN router config that was posted in the Member's area.  Hopefully this is still the correct one.




    Internetwork Expert - The Industry Leader in CCIE Preparation
    http://www.internetworkexpert.com

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  • @aquabrown - No worries mate. Just want to make sure the INE folks know to update both step 7 and 8.

    @mark - From what I can tell the CSS provided in your response for Step 8 is correct.  Calls towards the PSTN are making it from CCM > r1-hq > PSTN router, but they are failing to ring the PSTN phone due to the following (debug q931 on PSTN router)

     

     

    Called Party Number i = 0xA0, '+13122012222

    Dec 15 16:08:23.579: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x8004 

    Cause i = 0x8281 - Unallocated/unassigned number

     

     

    So, the cause code for call setup failure on PSTN router is 'Unallocated/unassigned number'.  If the call is sent to the PSTN router without the + symbol the PSTN phone rings. Does something need to be different on the PSTN router compared to the previous 3.x tasks to support + in the called number and ring the PSTN IP phone (7971)?  My PSTN router config is from the Members area and all the default configs are dated August 11.  Just want to make sure there isn't something missing.  The default PSTN router config from the member's area is essentially a slim CME config with one IP phone and 6 ephone-dn's. 

     

     

     











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    Hey Mark,

     

    You are correct, the + is not accepted as a valid digit in the DNIS
    (Called Number) field for the gateway. The CSS on the gateway for the Called
    Party Transformation CSS should be CSS_HQ_CALLED_OUT which should contain PT_HQ_CALLED_OUT
    which should be assigned to a Called Party Transformation Pattern of +.! With Digit
    Discard set to PreDot.

     

    This way all calls sent to the gateway will have the + stripped from the
    called number.

     

    HTH

  • Thanks everyone, I am back on track now. I rolled back my VM's and went through the entire lab again and completed everything. For those of us using our own lab equipment there are some uncertainties going into this task because it doesn't start where the previous task left off. Overall I would say this is a great, but challenging, task if you have never worked with + before.  However, it warrants further reading in my opinion.  There are 3 guides that include excellent information. UCM7 SRND provides a good overview, UCM7 Features and Services Guide and UCM7 System Guide provide excellent technical explanations.  

  • Hey Mark,

    Take a look at these two blog articles I put out to help you with + dial call routing:




    Mark Snow
    CCIE #14073 (Voice, Security)
    Instructor
    Internetwork Expert

    On Dec 15, 2009, at 21:35, MarkH <[email protected]> wrote:

    Thanks everyone, I am back on track now. I rolled back my VM's and went through the entire lab again and completed everything. For those of us using our own lab equipment there are some uncertainties going into this task because it doesn't start where the previous task left off. Overall I would say this is a great, but challenging, task if you have never worked with + before.  However, it warrants further reading in my opinion.  There are 3 guides that include excellent information. UCM7 SRND provides a good overview, UCM7 Features and Services Guide and UCM7 System Guide provide excellent technical explanations.  




    Internetwork Expert - The Industry Leader in CCIE Preparation

    http://www.internetworkexpert.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx
  • It's not just because you have your own equipment.....INE doesn't do the "config files for each exercise" bit so it's awkward for everyone (I'm not complaining, btw).  It would have been nice to be its own "section" but that wouldn't have fit with the way the guide is organized.

    Anyway, as I mentioned, the way I did it was by doing the first 6 tasks as one lab and then a different lab session later on *starting* with 3.7. So I didn't have to worry about "leftovers" getting in the way of my execution/understanding of the task.

    Glad to see Part 2 of the dial plan series up as well [Y]

  • So, I rolled back my VM's and I'm going through several 3.x tasks again to work on my speed.  One thing I didn't note about task 3.7 before rolling back is what CSS should r1 and r2 be associated with?  I'm stuck, where VoIP to PSTN works but not PSTN to VoIP.  I know it's an oversight on my part.

    The task itself is quite hefty, but since it doesn't pick up where 3.6 left off it feels like in addition to the task we also need to take a stab at what else needs to be undone without any real sanity check to make sure we didn't undo too much to the point the task will never work.  Am I making sense?  Would it be possible to have a short addendum for configuration modification going into the task?  

     

    Device > Gateways > r1-hq, T1PRI > Call Routing Information - Inbound Calls

    Significant Digits = 4

    CSS <NONE>

    AAR

    PREFIX

     

     

     

  • The CSS needs to be able to see the internal partitions. I don't have my WB with me at the moment but make sure the CSS you use includes the internal lines and test from there.

  • Ok, it's all sorted out thanks to Josh. [Y]

    r1-hq T1 port needs to be assigned to CSS_HQ_DEVICE (assuming phones are assigned to PT_HQ_DEVICE)

    r2-br1 T1 port needs to be assigned to CSS_BR1_DEVICE (assuming phones are assigned to PT_BR1_DEVICE)

    After save/reset, issue no mgcp/mgcp commands on each gateway as stated in troubleshooting section.  

    At first I didn't realize I needed to assign the gateways to CSS_HQ_DEVICE and CSS_BR1_DEVICE.  I should have realized it much sooner. But, even when I did assign CSS I did not initiate the mgcp commands on the routers immediate after which is why I ran into so many instances of it partially working or in some cases, worked on one site but not the other.

     

  • That's the one!

     

    Like I said above, you're home free once the call reaches the gateway. Now as long as it can "see" the destination phone, the call will make it all the way. Whatever partition the phones are in should be in the CSS you use for the gateway.

     

    This saga has the added bonus of burning that no mgcp/mgcp tango into your fingers! [8-|]

     

     

  • Just to understand this conversation.. The configs in the book are correct? Please Advise

  • The configuration in the workbook along with the comments here regarding the CSSs applied to the gateway will work for you. If you have any question, you can e-mail me directly, and I'll be glad to help.

     

    Thanks

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