dial-peer based on source

[dan.letkeman]

Hello,

I have setup a sip trunk to a provider on our voice gateway, and I wanted to use it for a certain group of phones for long distance only, and have everyone else use the t1.

So i'm wondering if there is a way with cucm or just on the h323 router to route calls out the sip trunk based on the source ip, or dn.  The phones are currently registerd to the cucm, and all use the same voice gateway.  I'm doing all of the call routing on the cucm cluster and sending it to the h323 gateway where there are various dial-peers based on numbers dialed.  eg:

dial-peer voice 100 pots

description 7 Digit Dialing

destination-pattern 9[2-9]......

port 0/0/0:23

forward-digits 7

etc....

But I want dn's 7001-7010 to go out the sip trunk which is terminated on the same voice gateway for numbers matching 11 digit long distance.

Dan.

[Senthil Kumar Sankar]

Hi

This can be achieved by Partitions and CSS in CUCM.

For example ;

Say if you already have a Route-pattern 9! in CUCM pointing o H323 gateway.

Create another duplicate 9! in CUCM and put that in a Partition (Eg : SIPTRUNK-PT) and point it to the same gateway where you want to send the calls, But you have add  a prefix(Eg : 8) under the "Calling Party Transformation" of the Routepattern Window.

In your gateway create 2 dial-peers like if the call come with the prefix 8 point to SIPTRUNK, for others point to your E1'

The DN's 7001-7010 will be in a different CSS (eg: SPECIAL-CSS) and add the parition (SIPTRUNK-PT) into it.

Also make sure that the remaining DN's are in Different CSS where the have access for normal 9! routepattern which doesnt have prefix.

I believe this would work for you,

Regards
Senthil

[dan.letkeman]

Hmm....there is no prefix command in a voip dial peer.

2811voice(config)#dial-peer voice 901 voip

2811voice(config-dial-peer)#p

2811voice(config-dial-peer)#p

2811voice(config-dial-peer)#p

2811voice(config-dial-peer)#p

2811voice(config-dial-peer)#p?

paramspace  pcm-dump      permission  playout-delay

preference  progress_ind

2811voice(config-dial-peer)#p

You can only use the prefix command in a pots dial peer.

Any other ideas?

[nitsinha]

I would like you to try this:-

On the CUCM,

1. Create a new route pattern (say 9.1[2-9]XX[2-9]XXXXXX) and put it in a partition say PT_LD_SIP.

2. Put the DN's 7001 to 7010 in a different CSS (say CSS_LD_SIP) which has access to PT_LD_SIP while all other DN's should have access to the normal LD route pattern that you currently have.

3. For the Long Distance(LD) route pattern that other phones have access to, do not strip the access code(9) at the CUCM while for this new route pattern in PT_LD_SIP partition, do a pre-dot digit strip and discard 9 at the CUCM.

4. Now at the gateway you should be having 2 dial-peers for LD calls as shown below:- 

     For normal DN's

     -------------------------

dial-peer voice 101 pots

description 11 Digit Dialing

destination pattern 91[2-9]..[2-9]......

port 0/0/0:23

forward-digits 11

For DN's 7001-7010

-------------------------------

dial-peer voice 102 voip

destination pattern 1[2-9]..[2-9]......

session target ipv4:

session protocol sipv2

Hope this helps

Regards

Nitesh

PS:Please rate helpful posts

[Senthil Kumar Sankar · Accepted solution]

Hi Dan,

The prefix you are supposed give in CUCM settings. by doing it you can achieve

Call from NormalDN's will hit the gateway like-------->9[2-9]......--------->Point this to POTS

Calls from 7001-7010 DN;s will hit the gateway like------>8[2-9].....--------Point this to SIP..

Example for 7 digit dialing Route patern settings in CUCM

Normal DN's Route Pattern : 9.[2-9]XXXXXX

             Under Discard Digit Settings select :  NoDigits (so the call will hit the gateway with 9 + 7digitnumber which will match POTS dial-peer)

7001-7010 DN's Route Pattern : 9.[2-9]XXXXXX

             Under Discard Digit Settings select : PreDot (so the 9 will be removed)

             Under Prefix Digits Settings add :8 ( so the call will hit the gateway with 8 + 7 digitnumber which will match the VOIP Dial-peer with SIP)

The above is just an example which i have given you may have different Route Patterns in your setting according to that you can change it.

Regards

Senthil

[dijohn]

You could alternatively play with incoming dial-peers and translation patterns

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00803f818a.shtml

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

/divin

PS:Rate useful posts!

[dan.letkeman]

Thanks, that looks like it will work very well.

Dan.