01-04-2010 11:44 AM - edited 03-18-2019 10:58 AM
Hello,
i am trying to run CUCM 6.1 and 3845 router with VIC-4FXO as my voice hardware.
i can dial out from ip phones to pstn, but i have a problem with inbound calls.
In gateway port detail configuration there is a field named "attendant DN". when i enter a valid & available
ip phone extention number in this field, the call is routed directly to the number and ip phone rings.
But i want to provide dial tone for ALL inbound calls from pstn, which the inbound calling party can dial
any number and route to desired destination based on route partitions.
Kindly help out
thanks in advance
Solved! Go to Solution.
01-04-2010 03:14 PM
i think
you are using MGCP
try this way:
will assume internal extensions consists of 4 digits
create a translation pattren
make the pattren for example 8.XXXX
in the partition section chose a partition that i included in the voice gateway CSS
in the CSS section chose a CSS that as access to the internal extension numbers
also make sure to check the box called provide dial ton
in the gateway DN put the number as 8 ( please make sure that 8 is not included as first digit of any other number in your CCM if yes then pick any other number no used as first digit)
in this way when someone call the call will be sent to number 8 and then a dial ton will be provided i put explicitly four XXXX to restricted to only 4 digits to be dialed )
good luck and let me know if it works or not
01-04-2010 01:30 PM
I think you can accomplish this using TCL scripts. Check out this document.
This should at least get you pointed in the right direction.
HTH
01-04-2010 03:14 PM
i think
you are using MGCP
try this way:
will assume internal extensions consists of 4 digits
create a translation pattren
make the pattren for example 8.XXXX
in the partition section chose a partition that i included in the voice gateway CSS
in the CSS section chose a CSS that as access to the internal extension numbers
also make sure to check the box called provide dial ton
in the gateway DN put the number as 8 ( please make sure that 8 is not included as first digit of any other number in your CCM if yes then pick any other number no used as first digit)
in this way when someone call the call will be sent to number 8 and then a dial ton will be provided i put explicitly four XXXX to restricted to only 4 digits to be dialed )
good luck and let me know if it works or not
01-05-2010 05:55 AM
thank you for your help.
actually i am using mgcp and i need to handle the problem in CUCM, so i can't use TCL script.
thanks to marwanshawi, by using translation pattern, my inbound calls hears a dial tone. but now i face another
problem for routing them. actually it is a TEHO scenario, which i must route inbound calls from a gateway,
to another gateway. but now after dial a number to second site/gateway, i hear "your call cannot be completed as dialed ..."
it might be a problem in partitions and CSS. here it is my config:
main-css inclides these partitions: (incoming-trunk, site1-gw, site2-gw, ip-phones)
partition | CSS | |
---|---|---|
MGCP gateway | main-css | |
translation pattern | incoming-trunk | main-css |
route pattern to site1 | site1-gw | |
route pattern to site2 | site2-gw |
what is wrong?
01-05-2010 03:24 PM
can you just draw your call flow i cant answer without understanding the call flow
thank you
01-05-2010 07:41 PM
okay
consider a call from person1 in site1 to person2 in site2.
the call initiated from pstn/pbx connected to the FXO ports of site1 gateway.
then the translation pattern provide the dial tone. then person1 dial 22.XXXX, which is routed by a route pattern to site2 gateway.
so the site2 gateway must dial out the XXXX number on its FXO port.
but it dosn't work because after entering 22XXXX cucm respondes by "your call cann't be completed as dialed ...."
thanks
01-05-2010 09:13 PM
what is the patrren you used
in the translation pattren ?
i think it has to be 8.22XXXX where 8 the digit that i told about above ( not sure what number you used )
i need to understand the second part
site1 voice GW---CCM ---translation--route-pattern 22.XXXX---site2 GW---FXO-- ( is it here PSTN phone or what??)
have you tried to send the cal direct from gateway to gateway
site1 gateway
dial 1234 to get dial ton
dial-peer voice 10 voip
dstination-patren 1234
session target
int site2
dial-peer voice 10 pots
destination-pattren 1234
port x/x/x
where x/x/x the FXO port
good luck
01-05-2010 10:10 PM
yes. before i move to CUCM, the solution was working fine based on route pattern configuration on router just like what you wrote above.
and also the translation pattern is 8.22XXXX, where 8 (PreDot) will be discarded.
then it matchs in route pattern 22.XXXX, which it routed to site2 gateway.
FXO ports of site2 gateway are connected to traditional PBX with 4 digit extensions like XXXX.
is problem because of CSS and partitions?
01-06-2010 01:46 AM
maybe
make sure that the CSS of the traslation contains the partition of the route pattren of 22.XXXX
also to to make sure its working dial from an irnal IP phone to the translation and to the route pattren directly ( also make sure the phone has the required CSS)
good luck
if helpful Rate
01-09-2010 12:45 AM
Dear
I read the CIPT 1 and CIPT 2 and also Cisco Cvoice , But I feel that I can't undrestand Cisco Call Manager 6.1 , do you have any other Document or resource that I can use as my reference for problems like this ?
Best Wishes
01-09-2010 01:08 AM
try this link:
Understanding and Using Partitions and Calling Search Spaces with Cisco CallManager
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094b53.shtml
Although this link talking about older version but the idea and concept are the same
also the bellow book is very useful in term of IP telephony design concepts
Cisco IP Telephony: Planning, Design, Implementation, Operation, and Optimization
http://www.ciscopress.com/bookstore/product.asp?isbn=1587051575
good luck
if helpful Rate
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