yesterday ....... i post the discussion about how i can reserve one analog telephone line for one of my user so whenever he make outgoing call his calls go through one particular analog line and never found busy tone .... in reply to my this query one guy tell me to use partitions and CSS ... my question is lets say if i create new partition name " test" and assign my user to this partition so he can only use it whenever he makes PSTN dialing so his call goes through this partition and reached my GW but my GW have five different analog line's how GW came to know that the first line is reserved for all of the calls coming from this user which is assigned to this partition.
i have CUCM 8.5 and GW is Cisco 2800 series ......... i dont want my user to dial any extra digit or PIN
You have to make him dial extra digit (even if one digit) so router 2800 can differentiate calls and send them to the appropriate FXO port. Also there is one thing, what's the protocol between CUCM and router2800???
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Create a voice translation rule and apply the same to the respective voice-port.
Note: you have to translate the calling party number.
Create new PA/CSS. associate route pattern and the designated user to it. inside route pattern configuration page scroll down to called party transformation section ( attachment below ), set prefix digit and lets say digit (1) for example.
thus you will have 2-call route pattern to h323 router, one route pattern with partition and prefix digit which will be forwarded to h323 router. inside your h323router you will have two different destination patterns. create two dial peers, one pointed to specific FXO port.
Hope that helps
I assume that this is MGCP gateway
Please, try to do the following to achieve your requirment:
Try by configuring as per the above suggestion, if you have any issue please quota me. This user will able to dial the digit directly without adding any prefix or extra digit.
I hope this will help you or will drive you to right direction.
This seems simple requirement, but not very easy to manage because of the fact that with H.323/SIP, you have distributed dial plans in CUCM and gateway, and both work independent of each other.
I see challenge here is to first identify the caller in gateway. Since 'incoming called-number' has highest priority in gateway while selecting inbound dial peer, all calls will be routed and destination port will selected on the basis of this dial peer, you requirement will never be achieved if you doing so.
You need to use 'answer-address' command in gateway to select inbound dial peer so that you have control over identifying the caller and apply desired policies.
There could be more than one ways to achieve your requirement but I see only the following as of now;
1. Don't use incoming called-number to select inbound dial peer.
2. Use answer-address command to select inbound dial peer.
3. Let say you want DN 101 should always use port 0/0/0 for outbound call. Create two inbound dial peers, first with answer-address as 101 and other should be configured to match all ANI's.
4. Use Class of Restrictions (CoR). Configure CoR as per following;