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CUCM SIP-Trunk Multiple Ports

I would like to know if it possible to create a SIP Normalization Script in CUCM that changes the Destination SIP port based on the phone thats making the phone.

 

  Current Issue:

 

   ITSP Provider requested that I send calls from different areas to the same SBC IP address but different port.

        Example:

                  Phone (A) makes a call

                                                                RP-RL-RG------> SIP Trunk(5060)----->CUBE(5060<-->5001)------>ITSP1

 

                  Phone (B) makes a call

                                                                RP-RL-RG------> SIP Trunk(5060)----->CUBE(5060<-->5002)------>ITSP1

 

I can do this on the CUBE but that would require to many dial-peer and sip profiles, is this possible in the CUCM to change sip port by Device-Pool?.

 

 

Or is there another I can match and send both calls via the same destionation dial-peer with different sip ports.?

 

 

Thanks.

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1 ACCEPTED SOLUTION

Accepted Solutions
VIP Super Bronze

Zakiab,You cant do this, its

Zakiab,

You cant do this, its simply impossible. Your signalling ports are defined not on the endpoints but on the B2BUA (CUCM or CUBE). On CUBE you have the most flexibility, as you can modify the signalling port based on your dial-peers. However as you have rightly pointed, you will need an unscalable amount of dial-peers to have this for each end point...So I suggest you tell your ITSP to change their design or switch to a new one.

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
4 REPLIES
VIP Super Bronze

Zakiab,You cant do this, its

Zakiab,

You cant do this, its simply impossible. Your signalling ports are defined not on the endpoints but on the B2BUA (CUCM or CUBE). On CUBE you have the most flexibility, as you can modify the signalling port based on your dial-peers. However as you have rightly pointed, you will need an unscalable amount of dial-peers to have this for each end point...So I suggest you tell your ITSP to change their design or switch to a new one.

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Hi Zakiab.Agree with my buddy

Hi Zakiab.

Agree with my buddy Deji (+5).

A way to achieve this could be , as usual, matching incoming calling number with the "answer-address" option on dialpeer and setting a prefix to force a call to match an outgoing dialpeer.

Eg

 

Calls coming from extension 1000

voice translation-rule 1

rule 1 /.*/ /1000\0/

dial-peer voice 1000 voip

incoming called-number .T

answer-address 1000

dial-peer voice 1100 voip

destination-patter 1000T

session-target ipv4:x.x.x.x:5062

 

 

etc etc

Obviously, that requires a lot of configuration in your cube, but it should work

 

HTH

 

 

Regards

 

Carlo

 

 

Please rate all helpful posts "The more you help the more you learn"

To add little to these two

To add little to these two great post +5

 

SIP normalization script will only work with ITSP if trunk is directly terminated on CUCM. But in your case there is a CUBE in between , so whatever changed you apply through script will only applicable to the call leg from/to CUCM and CUBE.

To modify SIP message you need to do changes on CUBE  towards ITSP on outbound dial-peer. You have two options either change your ITSP as Deji suggested or use Carlo's method or as i was suggested in your previous post regarding SIP profile.

 

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Thanks

Manish

Thanks Carlo. The current

Thanks Carlo.

 

The current method is that we match incoming calls from CUCM using URI

dial-peer voice 989 voip
 description "From CUCM"
 translation-profile incoming PREFIX
 session protocol sipv2
 incoming uri via 3000
 voice-class codec 1

voice class uri 3000 sip
 host ipv4:10.177.1.5(CUCM IP)


voice translation-rule 1
 rule 1 /^(100*)\/ /80\1/
 rule 2 /^(200*)\/ /90\1/

voice translation-profile PREFIX
translate called 1

 

The current method is that we match OUTGOING calls TO ITSP using SIP Profiles. 
------------------------------------------------------------------------------------
voice translation-rule 2
 rule 1 /^80(.*)\/ /\1/
 rule 2 /^90(.*)\/ /\1/

voice translation-profile REMOVE-PREFIX
translate called 2

voice class sip-profiles 200
 request INVITE sip-header Via modify ":5060;" ":5021;"


dial-peer voice 9 voip
 description "TO ITSP-90"
 voice-class sip profiles 100
 destination-pattern 80.T
translation-profile OUTGOING REMOVE-PREFIX
 session protocol sipv2
 session target ipv4:10.88.78.15:5021
 voice-class codec 1

 

voice class sip-profiles 200
 request INVITE sip-header Via modify ":5060;" ":5020;"


dial-peer voice 9 voip
 description "TO ITSP-90"
 voice-class sip profiles 200
 destination-pattern 90.T
translation-profile OUTGOING REMOVE-PREFIX
 session protocol sipv2
 session target ipv4:10.88.78.15:5020
 voice-class codec 1

 

 

 

Also I would like to know if it possible to change the default sip ports on the incoming dial-peer using sip-profiles and for that port to be used in the destination profile per the port change in the incoming dial-peer.?

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