Globalization & Localization Issue

[anis_cisco]

Hi all,

In my lab i have H.323 GW connected to PSTN. I configured Localization & Globalization& its working perfect.

At ip phone in miss call directories its showing me E.164 number include + sign. i want to dial from miss call directory without using edit key.

I go through with cisco document & they mentioned to achieve this you need to make route pattern for split + character.I did that but its not working.

In my scenario, when i call from PSTN to ip phone, on ip phone it display as 07978652817 & if i didnt answer call it go to miss call, in miss calls tab its showing +447978652817. As i press dial from miss call directory its not dialing.

I made Route-Pattern as \+44.7978652817, having partition which is in IP Phones CSS, with Discard Digit as Pre-Dot.

And On GW side i made dial-peer as

dial-peer voice 6 pots
destination-pattern 7978652817
port 0/1/0:23
forward-digits 10

Im trying since last two days, and im not successfutll, i dont know what & where im doing mistake. I tried to enable debugs " debug voice ccapi inout"on GW but nothing is coming there, can any one tell me how to troubleshoot it ???

any ones help will be greatly appreciated.

[iptuser55]

you need to add a 0 to the number so it becomes  07xxxx or . Either add it in CUCM and change the Dial-Peer to dest-patt 077xxxxxx or add it as a prefix 0 on the Dial-Peer

[anis_cisco]

Hi Bro,

You mean, my route-pattern should look like this

\+44:07978652817

any my dial-peer should look like this

dial-peer voice 6 pots
destination-pattern 07978652817
port 0/1/0:23
forward-digits all

???

I tried above but still when i call from miss call directory it shows +447978652817 on fone display & gives fast busy tone !!!

If you want i can share the snap shots + GW configuration ??

Regards

[iptuser55]

Yours

+44:07978652817

any my dial-peer should look like this

dial-peer voice 6 pots
destination-pattern 07978652817
port 0/1/0:23
forward-digits all

???

It should be

RP

\+44.7978652817            if your RDP pattern ibegins as +447......

predot Discard

Prefix 90

Send to GW 907978652817

Dial-peer voice 6 pots

dest-patt 9T

port 0/1/0:23

or

Dial-peer voice 6 pots

dest-patt 907978652817

forward 11

port 0/1/0:23

Basicallly we shoud send 90 to the GW, In the UK the leading digit 0 is the same as in the US prefix of 1 - 91212....... , or 1[2-9]xx[2-9]xxxxxx etc

[anis_cisco]

Still the same bro

any other suggestion ??

Regards,

[iptuser55]

Ok

Do a DEBUG ISDN Q931 on the GW - does the call get there check the called number- does it tally with the Dial-Peers?

Next need to check the \+44 route pattern, are you doing any XFORMS on the called Number?.

Since you can not dial + via the key pad - or can you jump to 2.  ????

If not

1. Create a Translation Pattern 9.4477!

Discard Predot

prefix  +

TP CSS - send it to the RP \+44777 - this will test the RP

2. Go to the route pattern \+447.... and set the Route config. to Block and set Precedure level exceeded.....

Make the call - It should fail with a Voice Annoucment - if it does then you know the RP \+44 is good up to here, check the GW  if it fails then check the CSS, Partitions of phone etc

let me know the outcome

Do a DNA as well

[anis_cisco]

Man,

Thats the problem, as i mention in my first post that im not able to see any logs on GW, if i could have got logs of debug isdn q931 or debug voice ccapi inout i would have sort out the problem before ... when i call from miss call directory, i m not able to see any logs on GW & this is the point where im facing difficulty to troubleshoot the problem.

I cant say i m doing mistake some where in GW bcoz if i create a RP e.g. 9.7978652817 ( which is number on PSTN Fone) with predot discard digit & RL of the GW & configure destination-pattern as 7978652817 in dial-peer @ GW, its working perfect ....

Only if i call a number +447978652817 from miss call directory its not showing me any debug on GW only fast busy tone with " +447978652817 " as a display on fone.

Im very thankfull for your time & help .

Regards,

[iptuser55 · Accepted solution]

Did you set the \+44 RP to block to check if it is getting even there? If not then has to be a CSS and partition issue

[anis_cisco]

Yah man i created and still the same busy tone i didnt get any voice announcement msg ...

Let me reset all config & try me again ....

Will let you know shortly

Thanks alooot

[anis_cisco]

Hi Man,

Just one quick confirmation, i m using CIPC 7.0 as aPSTN Phone ?? because i read it some where that CIPC doesn't support +characters ???

As we can call to all other numbers at PSTN except the numbers which are starting from + Character ???

Regards,

[anis_cisco]

Hi man,

I made it worked,

You were right i was doing mistake in assigning the CSS & Partitions, but i didnt understand the way it worked :

I already had a CSS with name CSS_SC & it contains parition PT_SC.

I assigned CSS_SC in GW for inbound, also i assigned CSS_SC in Device Pool for calling party transformation.

I assigned  PT_SC at Calling Transformation Pattern. Till yet i was able to see 8 digit on ip phone & E.164 in miss calls directory.

I made a new CSS with name CSS_Globlize & Assign only one parition PT_Globalize in it.

I assigned PT_Globlize in Route Pattern & I assigned CSS_Globlize to my ip phone. With this it worked ...

Can you tell me the procedure & steps for the selection of Partitions & CSS for globalization & localization ???

Thanks for your time & support

Regards,

[iptuser55]

Globalization

On the GW page half way down you see a section for Incoming number prefix for each type Subscriber, National, International and Unknown

Create a pattern based upon the Incoming number view via DEBUG ISDN Q931

So if your area code is say 212 and if national number come in without the leading digit 1 and local are 7 digits long then

1. Subscriber +1212
2. National +1
3. International +

Now calls will be seen on the phones as the +1212-local and National , + for international

Localized

Best to separate out the PT and CSS for XFORMS and normal PSTN access

• Create a new  PT called PT-SITENAME or AREA CODE -XFORM-ANI

CSS called CSS-XFORM-SITENAME or AREA CODE -XFORM-ANI - Can assign it to the users phone- be aware of any tick box referencing the DP or to the Device Pool

• Go to Transformation>>>Calling Transformations ( INBOUND TO THE USER,PHONE)

For local calls i.e 1212

\+1212.!

Partition- PT-SITENAME or AREA CODE -XFROM-IN

Predot discard

The above capture will capture calls, ANI being delivered to the phone. If you transfer or you dial TEHO you need extra setting on the Calling Transformation but the above is the basics

Oubound

Route Pattern

\+1.!