ciscoMeetingServer 2.1 and Outbound Calls

[maqsood ahmed]

Dear experts !

i have a situation in  CMS 2.1 , outbound calls.  i need to route the external calls from CMA / webClient   to SIP Gateway. 

i see a invite coming into Sip Gateway with calling name uri. as shown below ,,, the SIP gateway will NOT route it to SIP ISP if it is not from the known DID number

INVITE sip:1238402@96.1.163.10:5060 SIP/2.0

From: "user1" <sip:user1@12.163.1.10>;

there is nothing i can do in SIP trunk connection from cucm to cms2.1..there is  no provision to change the calling name uri to number

format in cucm.

how can i change the ( calling name)   <sip:user1@12.163.1.10>;   to something like   <sip:852741@12.163.1.10>   in   CMS 2.1.

[Adarsh Chauhan]

Hi,

I understand that you are asking for a configuration on CMS, but what about applying a sip-profile on the gateway.

example:

request ANY sip-header From modify "<sip:user1@12.163.1.10>" "<sip:852741@12.163.1.10>"

Before:

INVITE sip:2222000020@9.13.40.250:5060 SIP/2.0
Via: SIP/2.0/UDP 9.13.40.249:5060;branch=z9hG4bK1A203F
From: <sip:user1@12.163.1.10>;tag=F11AE0-1D8D
To: <sip:2222000020@9.13.40.250>
Date: Mon, 29 Oct 2007 19:02:04 GMT
Call-ID: 4561B116-858811DC-804DEF2E-4CF2D71B@9.13.40.249
Cisco-Guid: 1163870326-2240287196-2152197934-1290983195
Content-Length: 290

After:

INVITE sip:2222000020@9.13.40.250:5060 SIP/2.0
Via: SIP/2.0/UDP 9.13.40.249:5060;branch=z9hG4bK1A203F
From: <sip:852741@12.163.1.10>;tag=F11AE0-1D8D
To: <sip:2222000020@9.13.40.250>
Date: Mon, 29 Oct 2007 19:02:04 GMT
Call-ID: 4561B116-858811DC-804DEF2E-4CF2D71B@9.13.40.249
Cisco-Guid: 1163870326-2240287196-2152197934-1290983195
Content-Length: 290

Please rate and mark correct if helpful

Regards,

Adarsh Chauhan

[maqsood ahmed]

thanks adarsh !

iam still not decided where i should implement it in  CUCM or  VoiceGateway !

i think  CUCM normalization script is good place ,but it needs to modified in multiple places.. like in  from and  contact  headers.   And i have never done it before !

currently iam trying to study the " A guide to SIP Normalization"  ,,,and pickup something to write ...

meanwhile  is the below script i tried to make is this good ... i need to change only the calling name into number

M = {}
function M.inbound_INVITE(msg)
    local orig_to = msg:getHeader("From")
    local new_to = orig_to:gsub("user1", "852741")
    msg:modifyHeader("From", new_to)
end
return M

or u  have better one !

regards/

[Adarsh Chauhan]

Hi Maqsood,

You can definitely go ahead with the lua script and get this done but i would prefer sip-profile on the gateway as its very flexible and you can change things on Fly in a very convenient way.

Also to give you a heads up LUA script is not TAC supported whereas SIP-profile is (if that's applicable to you in any way).

I cannot comment on the LUA script as I haven't implemented it myself. 

Please rate and mark if helpful

Regards,

Adarsh Chauhan

[maqsood ahmed]

hi adarsh !

the below  profile changes  what i need ,, but  it also changes if the call is coming from cucm as well ,,, effectively all calls will have 13301234  as the calling number !

but ,,,i need it should look at  @example.com ,,if it found this domain ,,then change the only the first part of uri !    user1@example.com ,,,,,change to 13301234@example.com

still working on it !

request INVITE sip-header From modify "(<.*:)(.*@)" "\13301234@"

[Ammar Saood]

y dont you import users using telephone number field. so they should have a numeric userid rather than alphanumeric.

synchronize username parameter in CMS to TelephoneNumber field in AD.

username  =  $telephoneNumber$@domain.com

HTH

AMMAR

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