Questions about dial peers, route patterns and translation profiles

sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
[FONT=&amp]Hi guys,[/FONT]

[FONT=&amp]Recently I came across the article about translation rules.
[/FONT]
[FONT=&amp]What if, for those same inbound calls, we want to modify the caller-id to prefix the access numbers (9, 91 or 9011) so that the calls can be dialed easily from a user’s call history? To accomplish this, we need another set of rules:[/FONT]
[FONT=&amp]voice translation-rule 2[/FONT]
[FONT=&amp]rule 1 /^…….$/ /9&/[/FONT]
[FONT=&amp]rule 2 /^[2-9]..[2-9]……$/ /91&/[/FONT]
[FONT=&amp]rule 3 /^1[2-9]..[2-9]……$/ /9&/[/FONT]
[FONT=&amp]rule 4 /.*/ /9011&/[/FONT]
[FONT=&amp]
[/FONT]
[FONT=&amp]Imagine that I am in Chicago and someone is calling me from Boston. Does it mean on the screen of my phone display I will see something like 91617XXXXXX? What is the purpose of having 91 being displayed in front of 617XXXXXX instead of just 617XXXXXX? [/FONT]
Best, sacredboy!

Comments

  • davenulldavenull Member Posts: 173 ■■■□□□□□□□
    When people see a calling number starting with "9" they know it's an external caller, as opposed to for example a 4-digit same site caller.


    People in the US are accustomed to see a local caller ID starting with 9 +10 (or 7) digits, a long-distance 91 + 10 digits, and international starting with 9011. Also depending on how the system is set up it is easier for user to dial from Missed Calls list if the number is already in a dialable format.


    This is done solely for user habits. With a globalized dial plan it would make it much more simple to set up if people got used to dialing and receiving numbers in only +E.164 format.
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi Evgeny,

    Thank you for your reply. There is another little gap with translation rule which I would to clarify. Here is a chunk of text from the same article.
    Another action your router can take is to properly format your internal users’ caller-id for calls outbound to the PSTN. We created rules like this in Part 1. Here is a ruleset for our three sites:
    voice translation-rule 3
    rule 1 /^7…$/ /7035557…/
    rule 2 /^3[0-4]..$/ /9725553…/
    rule 3 /^5[23]..$/ /4065559…/
    And here is the voice translation-profile that has the router use voice translation-rule 3 to modify caller-id:
    voice translation-profile PSTN-OUT
    translate calling 3
    dial-peer voice 9 pots
    destination-pattern 9[2-9]..[2-9]……
    translation-profile outgoing PSTN-OUT
    port 0/0/0:23
    This I suppose is used when the WAN is down and call flows can't go through WAN circuit. In this example dialed 7XXX/3XXX/5XXX translate into 7035557XXX/9725553XXX/4065559XXX respectively. But what about 9 for access the trunk and 1 for national dialling (if necessary)? Does translation-profile outgoing PSTN-OUT works together with destination-pattern 9[2-9]..[2-9]……?
    Best, sacredboy!
  • davenulldavenull Member Posts: 173 ■■■□□□□□□□
    In this example dialed 7XXX/3XXX/5XXX won't translate into 7035557XXX/9725553XXX/4065559XXX. This translation profile only changes calling number, not called.


    In real life the calling number is usually already in the correct format, taken from External Phone Number Mask field in CUCM, but it can also be manipulated in several other places in CUCM.


    I'm not sure what your concern is about 9. Notice it's a pots dial-peer and that 9 is going to be stripped off.
  • bobfromfplbobfromfpl Member Posts: 104
    Hello Sacredboy,

    The translation pattern will only be applied AFTER the destination pattern has been matched. For instance, if a call is received at the gateway with the called number of 92123435656, then the voice gateway finds a match on dialpeer 9 and will then process the digit string through the translation profile. In your scenario it will modify the calling (ANI) number into a proper format before sending the call out the PSTN. As davenull mentioned, it will strip the leading 9 because its a POTS dialpeer. Likewise, you will need an additional dialpeer if you want to match calls that start with a '91'.
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi guys,Could you please help me to solve this task. Below is a diagram.[LONDON 3333 XXXX]
    E1 0/0/0:15 [PSTN simulator] T1 0/1/0:23
    [555 XXXX NEW YORK]I configured some dial-peers and translation profiles:
    voice translation-rule 1/\(^00\) \(.*\)/ /\2/
    voice translation-profile PSTN
    translate called 1
    voice port 0/0/0:15
    translation-profile incoming PSTN
    dial-peer voice 1 pots incoming called-number .
    direct-inward-dial
    
    dial-peer voice 2 pots 
    destination-pattern 212555....
    port 0/1/0:15
    forward-digits all
    
    1. In London someone is dialing 0012125555555 that is accepted by PSTN accoding to dial-peer voice 1 pots.
    2. The number 0012125555555 is matched to voice translation rule 1 and is changed to 12125555555.
    3. In the task, according to dial-peer voice 2 pots the number 2125555555 should come out of voice port 0/1/0:15 and go to router NEW YORK.

    The thing is that after translation rule 1 is processed I have 12125555555, but not 2125555555.

    The question is what should I do with 1 from the number 12125555555? Of course, I can change dial-peer voice 2 pots and make it like this:
    destination-pattern 1212555....
    forward-digits all
    
    However, if I add national calls in voice translation rule 1, then the number 12125555555, let's say from Chicago, translated into 2125555555, but it won't match to dial-peer 2 which, after we edited it, sends 1212555....
    voice translation-rule 1
    /\(^011\) \(.*\)/ /\2/ <= international
    /\(^1\) \(.*\)/ /\2/ <= national
    
    1. How this issue can be solved?
    2. Imagine, a company buys DID 212555XXXX. In this case telephone provider send 212555XXXX or 555XXXX?
    Best, sacredboy!
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi guys,

    Here are my translation rules and dial peers on PSTN simulator.
    voice translation-rule 1 rule 1 /\(^0011\)\(.*\)/ /\2/
     rule 2 /\(^0\)\(.*\)/ /\2/
     rule 3 /^........$/ /&/
     rule 4 /\(^011\)\(.*\)/ /\2/
     rule 5 /\(^1\)\(.*\)/ /\2/
     rule 6 /^.......$/ /&/
    !
    voice translation-rule 2
     rule 1 /^2\(.*\)/ /\1/ type any subscriber
     rule 2 /^8/ /&/ type any national
     rule 3 /^61/ /&/ type any international
    !
    voice translation-rule 3
     rule 1 /^212\(.*\)/ /\1/ type any subscriber
     rule 2 /^312/ /&/ type any national
     rule 3 /^1/ /&/ type any international
    !
    voice translation-profile NEW-YORK
     translate calling 3
    !
    voice translation-profile PSTN
     translate called 1
    !
    voice translation-profile SYDNEY
     translate calling 2
    !
    voice-port 0/0/0:15
     translation-profile incoming PSTN
     translation-profile outgoing SYDNEY
    !
    voice-port 0/1/0:23
     translation-profile incoming PSTN
     translation-profile outgoing NEW-YORK
    !
    dial-peer voice 1 pots
     incoming called-number .
     direct-inward-dial
    !
    dial-peer voice 2 pots
     description **SYDNEY 23333XXXX**
     destination-pattern ^6123333....
     direct-inward-dial
     port 0/0/0:15
     forward-digits 9
    !         
    dial-peer voice 3 pots
     description **NEW YORK 212555XXXX**
     destination-pattern ^1212555....
     direct-inward-dial
     port 0/1/0:23
     forward-digits 10
    !
    ephone-dn  1
     number 000
    !
    ephone-dn  2
     number 911
    !
    ephone-dn  3  dual-line
     number 61211111111 secondary 211111111
    !
    ephone-dn  5  dual-line
     number 12128888888 secondary 2128888888
    !         
    ephone-dn  6  dual-line
     number 13129999999 secondary 3129999999
    
    I have one interesting thing with international calls both on CUCM on the one side and CUCME on the other side. The thing is that if international dial-peer on CUCME is configured like this:
    dial-peer voice 12 pots description **INTERNATIONAL DIALING**
     destination-pattern 9011T
     port 0/0/0:23
     [B]prefix 011[/B]
    
    And Route Pattern for international calls is configured like this:
    0.0011!
    
    Then international calls don't go out. However, if I explicitly specify the number of digits that should be forwarded and change prefix 011 for forward-digits 11 on CME and change 0.0011! for 0.0011XXXXXXXXXXX on CUCM then international call flows work perfectly.

    Why is it so?
    Best, sacredboy!
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi guys,

    I have the following voice translation rule and dial peers:
    voice translation-rule 3
    rule 1 /^\(3...\)$/ /6123333\1/
    !
    voice translation-profile PSTN-OUT
    translate called 3
    !
    voice-port 0/0/0:23
    translation-profile outgoing PSTN-OUT
    !
    dial-peer voice 3000 voip
    description **OUGOING CALLS TO SYDNEY OVER WAN**
    destination-pattern 3...
    session protocol sipv2
    session target ipv4:192.168.10.5
    !
    dial-peer voice 3001 pots
    description **OUGOING CALLS TO SYDNEY OVER PSTN**
    translation-profile outgoing PSTN-OUT
    preference 1
    destination-pattern 3...
    port 0/0/0:23
    prefix 9011
    
    NY-RTR#test voice translation-rule 3 3002
    Matched with rule 1
    Original number: 3002 Translated number: 61233333002
    Original number type: none Translated number type: none
    Original number plan: none Translated number plan: none
    
    I need 3ХХХ is automatically translated into 901161233333002 (9 - trunk code; 011 - international access code; 61233333002 country code + area code + number) in case WAN link is failed.

    For some reasons it doesn't work. I disconnect WAN cable and try dial 3ХХХ, but it is not translated into 901161233333002.

    Could you please check if voice translation rules and dial peers are configured properly?
    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Hi guys,

    I have the following voice translation rule and dial peers:
    voice translation-rule 3
    rule 1 /^\(3...\)$/ /6123333\1/
    !
    voice translation-profile PSTN-OUT
    translate called 3
    !
    voice-port 0/0/0:23
    translation-profile outgoing PSTN-OUT
    !
    dial-peer voice 3000 voip
    description **OUGOING CALLS TO SYDNEY OVER WAN**
    destination-pattern 3...
    session protocol sipv2
    session target ipv4:192.168.10.5
    !
    dial-peer voice 3001 pots
    description **OUGOING CALLS TO SYDNEY OVER PSTN**
    translation-profile outgoing PSTN-OUT
    preference 1
    destination-pattern 3...
    port 0/0/0:23
    prefix 9011
    
    NY-RTR#test voice translation-rule 3 3002
    Matched with rule 1
    Original number: 3002 Translated number: 61233333002
    Original number type: none Translated number type: none
    Original number plan: none Translated number plan: none
    
    I need 3ХХХ is automatically translated into 901161233333002 (9 - trunk code; 011 - international access code; 61233333002 country code + area code + number) in case WAN link is failed.

    For some reasons it doesn't work. I disconnect WAN cable and try dial 3ХХХ, but it is not translated into 901161233333002.

    Could you please check if voice translation rules and dial peers are configured properly?

    apply the voice translation profile to the dial-peer not the voice-port. that's best practice; might have another dial-peer going out that voice port and that might mess things up. Also, testing the "voice translation rule" won't take into account the prefix you used for dial-peer 3001 (prefix 9011)...it runs the test solely on the translation rule not considering what dial-peers reference it. do the prefix on the voice translation-rule instead and run a test then. before you do this though, try doing a "no digit-strip" on your POTS dial-peer first...might get it working with the way you have things set up now..but best practice is to apply the translation-rules/profiles to the dial-peers.

    this might help: http://what-when-how.com/wp-content/uploads/2012/03/tmp1D236.jpg
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Thank you negru_tudor,

    Just got another one question. How do I do the same but this time on CUCM?

    Thank you.
    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Thank you negru_tudor,

    Just got another one question. How do I do the same but this time on CUCM?

    Thank you.

    OK, first thing you'll need to understand is how CUCM routes calls and that is:

    Route Pattern -> Route List (Route Group1, Route Group2 etc)-> Route Group (Device1, Device 2 etc)

    Then you'll have to read up on "Calling/Called Party Transformations".
    CUCM uses these patterns to modify either the Called or Calling number (or both), exactly like a router uses the voice translation-rules.

    The difference here is that CUCM assigns these "transformations" into a Partition. You can basically bundle several transformations (called or calling) inside a partition.

    The way you use them however is you create a Calling Search Space (know as Called/Calling Party Transformation CSS) which acts similar to your router's voice translation-profile...since they're in a Partition it only makes sense to use a CSS to call them up :)

    Now, for your scenario let's assume you have Route Pattern 3xxx which can break out either via a SIP trunk (prefered option) or via an ISDN link hosted on a H323 or MGCP gateway. When it breaks out over the ISDN link it will have to convert the called number of 3xxx to 1-212-555-3xxx.

    Next, you create 2 Route Groups: one for the SIP trunks (best practice to bundle WAN links in the same group and TdM in their own), let's say it's RG-WAN, and one for the ISDN link(s), let's say it's RG-ISDN.

    The only member of the RG-WAN group will be your SIP trunk while in the RG-ISDN one, you only allocate the ISDN gateway.

    Then, you create a Route List, say it's named RL-Calls. This route list will dictate how your 3xxx calls are routed in between RG-WAN and RG-ISDN. Putting the RG-WAN first and RG-ISDN under will always try to route calls over the SIP trunk - this is how you need to put them for your scenario.

    MAKE SURE YOUR 3xxx ROUTE PATTERN POINTS TO THIS ROUTE LIST!

    Now, create a partition called "PT-PSTN-Called-OUT" and a calling search space called "CSS-PSTN-Called-OUT". (You'll see why, keep reading)

    Next, you create a Called Party Transformation Pattern, matching dialed digits 3xxx and pre-pend the 1-212-555 string to it. It will be mandatory to specify a partition so use "PT-PSTN-Called-OUT".

    You can either use the "Prefix" field or the "Transformation Mask" - try them both, toy around with it a little.

    What this will do is when CUCM sees calls going (called number) to 3xxx it will put the 1-212-555 prefix in front.

    But you don't want that to happen to calls going over the SIP trunk...this is where the "CSS-PSTN-Called-OUT" calling search space comes in...only devices (trunks, gateways etc) that have "CSS-PSTN-Called-OUT" assigned as their "Called Party Transformation CSS" will have access to this partition and thus to your 3xxx called party transformation pattern.

    Last phase is to go to the ISDN/H323 gateway and look for the "Outbound Calls" section.

    There should be a "Called Party Transformation CSS" drop down menu there.

    Click on that drop-down menu and you'll see your CSS-PSTN-Called-OUT search space and select it.

    What this now does is when calls go OUT through this ISDN gateway, CUCM will always check the CSS-PSTN-Called-OUT search space for a 3xxx CALLED NUMBER pattern.

    If it sees something that matches, it sticks the 1-212-555 mask/prefix to it so you get your PSTN-routable 1-212-555-3xxx number.

    You do the same for Incoming calls (ie. striping the called number from 1-212-555-3xxx to 3xxx) but under the "Inbound Calls" section of your ISDN gateway in CUCM...your Called Party Transformation Pattern will be 12125553xxx and you can match it like this 1212555.3xxx and use the Discard Digits: PreDot option IIRC.

    Hope this helps. It's a key topic in CIPT2 which leads to dialplan normalization which is/was a beast of a topic for CCNP Voice.

    Good luck.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi,

    Thank you very much for this broad explanation. :)

    I am trying to configure Called Party Transformation Pattern and stuck on this step. I'm getting the error on screenshot.

    Also, should I put 1212555 as a prefix or 000111212555 (0 - trunk code and 0011 - international dialing code)?
    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Hi negru_tuder,

    Thank you very much for this broad explanation. :)

    I am trying to configure Called Party Transformation Pattern and stuck on this step. I'm getting the error on screenshot.

    Also, should I put 1212555 as a prefix or 000111212555 (0 - trunk code and 0011 - international dialing code)?

    Hey bud,

    Try giving what I wrote down another read and go through these steps as you read.

    The error you got is because instead of configuring the 3xxx Called Party Transformation Pattern, you were trying to configure the partition.

    You first configure the partition like any other normal partition, then the CSS like any other normal CSS.

    THEN you configure the Called Party Transformation. Pattern: 3XXX, Partition: PT-PSTN-Called-OUT.

    About the prefix digits (access code or not), think about this a little bit...your call is going out to the PSTN. What will the PSTN / carrier do if you send the access code along? ..I'll tell you, it will drop the call... BUT this is a lab so it depends on what the PSTN router expects / is able to route.

    Remember, if you get stuck at any point, try backtracking a little bit, go through the details again and most importantly break stuff down into smaller manageable bits. Oh, and don't forget to rate useful posts :)
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    The error you got is because instead of configuring the 3xxx Called Party Transformation Pattern, you were trying to configure the partition.

    You first configure the partition like any other normal partition, then the CSS like any other normal CSS.

    THEN you configure the Called Party Transformation. Pattern: 3XXX, Partition: PT-PSTN-Called-OUT.
    Howdy mate ;),

    I am a little bit confused. Here is what you advised me to do earlier:
    Next, you create 2 Route Groups: one for the SIP trunks (best practice to bundle WAN links in the same group and TdM in their own), let's say it's RG-WAN, and one for the ISDN link(s), let's say it's RG-ISDN.
    I created two Route Groups; one with SIP Trunk only and another one with MGCP Gateway only.
    MAKE SURE YOUR 3xxx ROUTE PATTERN POINTS TO THIS ROUTE LIST!
    My 5XXX (yes it's 5XXX not 3XXX but it doesn't really matter) :).
    Now, create a partition called "PT-PSTN-Called-OUT" and a calling search space called "CSS-PSTN-Called-OUT". (You'll see why, keep reading)
    I got one Partition and one CSS.
    Next, you create a Called Party Transformation Pattern, matching dialed digits 3xxx and pre-pend the 1-212-555 string to it. It will be mandatory to specify a partition so use "PT-PSTN-Called-OUT".

    You can either use the "Prefix" field or the "Transformation Mask" - try them both, toy around with it a little.
    So what I'm doing next is proceed Call Routing -> Transformation -> Transformation Pattern -> Called Party Transformation Pattern. I assume this what you meant. I attached the creenshot of my Route Pattern 5XXX and another error I get when I try to create Called Party Transformation Pattern.

    Could you please have a look on it.

    Thanks. :)
    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Howdy mate ;),

    I am a little bit confused. Here is what you advised me to do earlier:

    I created two Route Groups; one with SIP Trunk only and another one with MGCP Gateway only.

    My 5XXX (yes it's 5XXX not 3XXX but it doesn't really matter) :).

    I got one Partition and one CSS.

    So what I'm doing next is proceed Call Routing -> Transformation -> Transformation Pattern -> Called Party Transformation Pattern. I assume this what you meant. I attached the creenshot of my Route Pattern 5XXX and another error I get when I try to create Called Party Transformation Pattern.

    Could you please have a look on it.

    Thanks. :)

    Hey man,

    The info I was trying to share was to help you get going on your way to routing calls; the 3xxx wasn't meant to be 100% accurate to what your lab really is like - more like a guideline for the sake of the example.

    OK, you need an ADDITIONAL partition and an ADDITIONAL calling search space. The ones you already have set up (SYD-International-PT and SYD-International-CSS) they're used for CALL ROUTING, as in this is where you stack and how you access your ROUTE PATTERNS (5xxx).

    ROUTE patterns are used for routing calls over trunks or gateways. TRANSFORMATION patterns are used to modify the Calling/Called numbers once the call has been routed out. The confusion here is that CUCM uses partitions and calling search spaces for both - that's what most people have a hard time getting their heads around when doing CUCM dialplans.

    CUCM won't allow you to have a route pattern and called party transformation pattern inside the same partition because it sees the same digits so conflict occurs (your "Add failed" error) - you get the same error if you try to assign 2 identical directory numbers into the same partition that's just how CUCM works.

    Called Party Transformation Patterns REQUIRE a separate partition and a separate calling search space. So, leave your existing routing as-is (route-patterns, route lists and route groups) but create:

    - 1 x new partition SYD-CalledOUT-PT
    - 1 x new calling search space SYD-CalledOUT-CSS

    Then create a Called Party Transformation Pattern of 5xxx and assign it to the SYD-CalledOUT-PT (don't forget your prefix magic).

    Last step, go to the MGCP gateway, and on the Outbound Calls section tick the "Called Party Transformation CSS" and select the SYD-CalledOUT-CSS.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi :),

    I created another Partition named SYD-CALLOUT-PT.

    I created another Calling Search Space named SYD-CALLOUT-CSS.

    I created Called Party Transformation Pattern with the following paarmeters:
    Pattern: 5XXX
    Partition: SYD-CALLOUT-PT
    (section) Called Party Transformations: Prefix Digits - 000111212555

    Then in Gateway Configuration, in the section Call Routing Information - Outbound Calls, in the Called Party Transformation CSS drop-down list I selected that SYD-CALLOUT-CSS.

    Unfortunately still not working. icon_sad.gif

    In this regards I got a question:
    Below is a the chunk of sh run | s voice translation-rule on PSTN simulator.
    voice translation-rule 1
    rule 1 /\(^0011\)\(.*\)/ /\2/
    I imagine this is exactly what PSTN router expects my MGCP gateway to send for international calls.
    3XXX [CUCM] --- [MGCP GTW] 61 2 33333XXX === WAN/PSTN === 1 212 5555XXX [CME] 5XXX (61 - Australia and 1 - USA).

    I tried both 000111212555 and 00111212555 but neither the first nor the second variant didn't work. The I noticed that in Called party Trasnformation Pattern Configuraton in the section Called Party Transformations the Discard Digit option is grayed and inactive (this can bee seen in my previos screenshot).

    Can it be the reason of the issue?

    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Hi :),

    I created another Partition named SYD-CALLOUT-PT.

    I created another Calling Search Space named SYD-CALLOUT-CSS.

    I created Called Party Transformation Pattern with the following paarmeters:
    Pattern: 5XXX
    Partition: SYD-CALLOUT-PT
    (section) Called Party Transformations: Prefix Digits - 000111212555

    Then in Gateway Configuration, in the section Call Routing Information - Outbound Calls, in the Called Party Transformation CSS drop-down list I selected that SYD-CALLOUT-CSS.

    Unfortunately still not working. icon_sad.gif

    In this regards I got a question:
    Below is a the chunk of sh run | s voice translation-rule on PSTN simulator.
    voice translation-rule 1
    rule 1 /\(^0011\)\(.*\)/ /\2/
    I imagine this is exactly what PSTN router expects my MGCP gateway to send for international calls.
    3XXX [CUCM] --- [MGCP GTW] 61 2 33333XXX === WAN/PSTN === 1 212 5555XXX [CME] 5XXX (61 - Australia and 1 - USA).

    I tried both 000111212555 and 00111212555 but neither the first nor the second variant didn't work. The I noticed that in Called party Trasnformation Pattern Configuraton in the section Called Party Transformations the Discard Digit option is grayed and inactive (this can bee seen in my previos screenshot).

    Can it be the reason of the issue?


    On the PSTN gateway do a "debug isdn q931" and look at what's coming in..post the debug here as well. If therr's no output your call might not be leaving CUCM (css and partitions issue)
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    working now?
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
  • sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Unfortunately not. I mean I tried to debug the issue using debug isdn q931 command but nothing was given to me in command line when I tried to dial a number.
    Best, sacredboy!
  • negru_tudornegru_tudor Senior Member Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Unfortunately not. I mean I tried to debug the issue using debug isdn q931 command but nothing was given to me in command line when I tried to dial a number.

    means that the call isn't leaving your cucm - mgcp gatway. check the isdn controller on the mgcp gateway (controller e1 0/0/1 for example or whichever it is) it needs to have a "pri-group timeslots 1-31 service mgcp" and then on the dial-peer on your mgcp gateway for the isdn link you need a "service mgcp"..

    you can also use the cucm dna (dialed number analyzer) - cucm will simulate a call flow and show you exactly how the call gets routed.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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