SIP Transparency and Normalization Questions

RE: CUCM Normalization to remove port 5060

2013-05-20T17:22:09Z

I personally haven't written a script to do it, but you should be able to accomplish this with SIP Normalization. You can use the 'getRequestLine()' function which returns the method, the request URI, and the protocol version. You can manipulate the request URI "[b][b][b]sip:6671234@10.10.10.10:5060[/b][/b][/b]" any way that you want. Looks like you want it to be: "[b][b][b]sip:6671234@10.10.10.10[/b][/b][/b]" You would then use the 'setRequestUri() function to put your new request URI in the INVITE. Mark

CUCM Normalization to remove port 5060

2013-05-18T13:23:42Z

Good day everyone as mentioned above, im looking to see if there is a way to remove port 5060 from the SIP messages coming out of CUCM. We are testing a topology where CUCM 8.6 is directly integrated with a SIP provider device. basically this is what they have caputured and recommeded REJECTED CALL FROM CUCM SIP/SDP request: INVITE sip:6671234@10.10.10.10:5060 - gets a 480 temporarily unavailable SUCESSFUL CALL SIP/SDP REQUEST: INVITE sip:6671234@10.10.10.10 - NO Port 5060 works any assistance would be greatly appreciated. thanks in advance

RE: SIP Normalization on Recording

2013-05-09T11:33:20Z

I have to start off by saying that I've never tried to use SIP normalization on a recording trunk, but I don't see any reason why it wouldn't work. To manipulate the header, you can start by determining quickly if the header for this message needs to be manipulated or not by using: local fromtag = msg:getHeaderUriParameter("From", "x-farendaddr") You can then retrieve the address from the header and check to see if it is ??????. If it is, you'll need to use 'modifyHeader' to change it's value; you can't update an existing Header URI Parameter (or remove & add it). That means you'll have to get the entire 'From' header by doing a: local fromheader = msg:getHeader("From") manipulate the string by substituting 'x-farendaddr=??????' with 'x-farendaddr=anonymous', then modify the header with the updated string: msg:modifyHeader("From", fromheader)

SIP Normalization on Recording

2013-05-09T04:56:42Z

Hi All, We currently have an issue on our recording platform having parsing error and not recording SIP calls with farendaddr=?????? (with UNKNOWN/PRIVATE calling ID). Would like to know how we can do about changing the whole address with ?????? to anonymous or 000000 since the recording platform only can support paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" unreserved = alphanum / mark mark = "-" / "_" / "." / "!" / "˜" / "*" / "’"/ "(" / ")" Below is a sample SIP Invite message that causes parsing error. sip:11120006678@170.241.177.66:5060 SIP/2.0 Via: SIP/2.0/TCP 171.194.204.20:5060;branch=z9hG4bK432b4772fca8d3 From: "Jeffrey Lim";tag=57345144~cd085c1c-4462-4066-8252-25989229b16e-54684584 Any form of help would be great. Regards, Ramon

RE: Insert attach data

2013-05-07T14:14:42Z

CUCM should transparently pass any 'unknown' headers through from one call leg to another. If the information is in a header that is 'known' to CUCM, it will consume the header (it is a B2BUA) instead of passing it transparently. Your transparency script has to make an unknown-header item out of the known header and put it back when it leaves CUCM in order to pass through transparently. This is described in the Developer Guide section on Transparency.

RE: Read Files from Normalization

2013-05-07T13:47:52Z

The Cisco SIP Lua environment only supports the string library in its entirety and a subset of the base library: For the base library, the following is supported: • ipairs • pairs • next • unpack • error • type • tostring

RE: Redirecting number mask with "CUCMs external phone number mask" ?

2013-05-07T13:45:01Z

You are correct that LUA script can only retrieve a parameter from the SIP Normalization configuration at the SIP trunk level. You could potentially have a different SIP trunk for each number range that needs a different external mask and define a different "mask" parameter value on the SIP Normalization setting on each of those SIP trunks.

RE: Redirecting number mask with "CUCMs external phone number mask" ?

2012-12-18T11:45:16Z

Prober format if msg(can't delete the syntax par): We need E164 number as redirecting number for outbound calls. External phonenumber mask needs to be applied to redirecting number as customer has approx. 50 number ranges. TACs only answer is LUA since CUCM can't be upgraded) CUCM is 8.6.1 (can't be upgraded at the moment since HCS) We found this bit on cisco, but as I read it it's retrieving the mask as a parameter from the LUA script, we need it to retrieve the relevant External phone number mask. -- M = {} local mask = scriptParameters.getValue("Diversion-Mask") function M.outbound_INVITE(message) if mask then message:applyNumberMask("Diversion", mask) end end return M

Redirecting number mask with "CUCMs external phone number mask" ?

2012-12-18T11:43:59Z

We need E164 number as redirecting number for outbound calls. External phonenumber mask needs to be applied to redirecting number as customer has approx. 50 number ranges. TACs only answer is LUA since CUCM can't be upgraded) CUCM is 8.6.1 (can't be upgraded at the moment since HCS) We found this bit on cisco, but as I read it it's retrieving the mask as a parameter from the LUA script, we need it to retrieve the relevant External phone number mask. -- M = {} local mask = scriptParameters.getValue("Diversion-Mask") function M.outbound_INVITE(message) if mask then message:applyNumberMask("Diversion", mask) end end return M

Read Files from Normalization

2012-12-06T17:54:54Z

Hi all, I have a list of names, to insert to my P-Asserted-Identity during the ringing state. It's working while I use it static at the Normalization body, but I have a big list of names for all my 200 DNs. I tryed to use dofile("./tftp/Oppa.lua") I put the file there, but its not working. Is this path correct? doFile work for CUCM? I will thy io.open next. Anyone can help me? Thanks

Insert attach data

2012-10-09T18:07:02Z

Hi ALL, My customer has a need to sending an information in attach data to the front end CTI OS of the Agent. And I don't know anything about programming. The flow are basically: PSTN VG -> IVR Dígitro -> SIP -> CUCM -> SIP -> CVP -> ICM -> Agent. The scripts are located in the IVR Digitro and the EC field (customer id) must appear on the front end CTI OS, so the IVR Dígitro needs to forward the attach data via SIP Header to CUCM. Right? I configured the Normalization in CUCM, and the test of CUCM to the CVP works, the information goes to the agent's screen correctly. I'm not able to do the CUCM make a bypass that attach data when the information comes from IVR Digitro. I understand that this case must to configure the transparency, right? I used the LUA script, but still is not working. What could it be? Thanks in advance.

CUCM - Nortel CS1000 SIP GW Integration Modify INVITE header

2012-09-28T19:01:27Z

Hi Team, I'm working in a CUCM-Nortel Integration, incoming calls from Nortel to CUCM fails due the received INVITE from Nortel is in this format: =========== INVITE sip:333666;phone-context=udp@test.com;user=phone SIP/2.0 . To: ============ CUCM rejects the call and in the trace I see the reason: |Digit Analysis: Host Address=test.com DOES NOT MATCH any address for this cluster. |Digit Analysis: Host Address=test.com DOES NOT MATCH top level org domain. It makes sense since the CUCM is not named test.com nor the top level org domain. I can fixed using the test.com in the CUCM>Enterprise Parameters> but is not really the solution I would prefer to use. I’m wondering if I can use the Normalization Script to grab the phone-context= tag from the To: header and rename it from [url=mailto:phone-context=udp@test.com]phone-context=udp@test.com[/url]; to phone-context=udp@x.x.x.x; where x.x.x.x is the ip address of the CUCM. I read the doc and I think I can use the following function to grab the tag M = {} function M.inbound_INVITE(msg) local to = msg:getHeaderUriParameter("To", "phone-context") end return M But how can be modified? Any help on this is really appreciated. Regards Carlito

Masking the Calling Name based on Calling number

2012-04-23T16:29:03Z

We are trying to write a normalization script to change the outbound calling name. We have several companies that are tenants as well, so changing it just on the trunk is not an option since all tenants utilize the same SIP trunk. We want to make the change based on the calling number. Here is the script that we have come up with, but it does not seem to be working. Also the trace is enabled on both the script and the trunk, but we are not seeing any trace in the sdi when we make calls. Any help or suggestions would be greatly appreciated. Bill M = {} trace.enable() local function modifyMessage(msg, newstring) local newstring1 = '"'..newstring..'"' trace.format("Newstring is %s", newstring) trace.format("Newstring1 is %s", newstring1) local paivalue = msg:getHeader("P-Asserted-Identity") local painewuri = newstring1.." "..string.match(value, "(<.+>)") trace.format("PAIValue is %s", paivalue) trace.format("PAINewuri is %s", painewuri) msg:modifyHeader("P-Asserted-Identity", painewuri) local fromvalue = msg:getHeader("From") local fromnewuri = newstring1.." "..string.match(value, "(<.+>)") trace.format("FromValue is %s", fromvalue) trace.format("FromNewuri is %s", fromnewuri) msg:modifyHeader("From", fromnewuri) local rpivalue = msg:getHeader("Remote-Party-ID") local rpinewuri = newstring1.." "..string.match(value, "(<.+>)") trace.format("RPIValue is %s", rpivalue) trace.format("RPINewuri is %s", rpinewuri) msg:modifyHeader("Remote-Party-ID", rpinewuri) end function M.outbound_ANY(msg) local from = msg:getHeader("From") if(from.find('636827...') not -1) then modifyMessage(msg, "Maritz") end if(from.find('6362262[0-5]..') not -1) then modifyMessage(msg, "Intellispend") end end return M

RE: Adding a "user" in "From" header ?

2012-01-20T15:13:44Z

HI BengT Anderson, I am an user and definitely not a developer. I ran into a similar issue as well, my sip uri looked like: I was able to get this routed the right way in call manager but not on my cisco voice gateway. What I did was to add a "SIP Route Pattern". I am running CM 8.6 and this is located in: http://callmanager/ccmadmin then go to; Call Routing > SIP Route Pattern. In my call flow, here is how I am setup: PSTN > 2pri > Voice GW > (h323) > CM What I did was that I created a sip trunk from CM to the voice gateway (cisco 2811) and then in the configuration of the sip route patern, I entered the IP address in the first field and in the bottom I selected the new sip trunk I created above and then CM started to route the calls the right way. THere might be an easier way of getting this done, I am not an expert. IF you happen to know how to get this done in the gateway please DO share though!..lol It would really help me on another project that I am working on.

RE: Adding a "user" in "From" header ?

2012-01-20T15:13:44Z

RE: New Message from Charles Ragan in Cisco Unified Communications Manager

2011-11-08T20:56:06Z

Rune ¿ if the pai is there (it would be the original To: I believe) ¿ you should be able to do something similar to... M = {} function M.outbound_INVITE(msg) local uri = msg:getUri("P-Asserted-Identity") msg:removeHeader("Diversion") msg:addHeader("Diversion", uri) end return M Rune Heggelund has created a new message in the forum "SIP Transparency and Normalization Questions": -------------------------------------------------------------- Thank you for replying. The setup is as follow; pstn-----sip----cucm-----sip------trio-vmail I can apply the script at the outgoing invite to the trio-vmail. I still need some help to create the script. Here is what sendt in the Invite header; Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes If I convert the diversion headers into History-Info headers for outbound INVITE messages. And then remove the Diversion header. This is what I got; M = {} function M.outbound_INVITE(msg) if msg:getHeader("Diversion") then msg:convertDiversionToHI() msg:removeHeader("Diversion") end end return M History-Info: ;index=1 History-Info: ;index=1.1 History-Info: ;index=1.1.1 The Trio System understands both diversion headers and History-Info headers. If I first convert the diversion headers into History-Info headers, then I got information I need in index=1. If I now remove all other index lines everything should be ok. -- To respond to this post, please click the following link: or simply reply to this email. -- To respond to this post, please click the following link: or simply reply to this email.

RE: New Message from Rune Heggelund in Cisco Unified Communications Manager

2011-11-08T20:28:06Z

Ahh..okay. So trio can¿t choose between 1st redirect or last redirect? ¿ In the values below ¿ what is the numbering scheme for the ¿nn¿ on the dh¿s? From: Cisco Developer Community Forums [mailto:cdicuser@developer.cisco.com] Sent: Tuesday, November 08, 2011 1:30 PM To: cdicuser@developer.cisco.com Subject: New Message from Rune Heggelund in Cisco Unified Communications Manager SIP (SIP) - SIP Transparency and Normalization Questions: RE: Delete the top-most diversion header or change order of Multiple Divers Rune Heggelund has created a new message in the forum "SIP Transparency and Normalization Questions": -------------------------------------------------------------- Thank you for replying. The setup is as follow; pstn-----sip----cucm-----sip------trio-vmail I can apply the script at the outgoing invite to the trio-vmail. I still need some help to create the script. Here is what sendt in the Invite header; Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes If I convert the diversion headers into History-Info headers for outbound INVITE messages. And then remove the Diversion header. This is what I got; M = {} function M.outbound_INVITE(msg) if msg:getHeader("Diversion") then msg:convertDiversionToHI() msg:removeHeader("Diversion") end end return M History-Info: ;index=1 History-Info: ;index=1.1 History-Info: ;index=1.1.1 The Trio System understands both diversion headers and History-Info headers. If I first convert the diversion headers into History-Info headers, then I got information I need in index=1. If I now remove all other index lines everything should be ok. -- To respond to this post, please click the following link: or simply reply to this email.

RE: Delete the top-most diversion header or change order of Multiple Divers

2011-11-08T19:30:09Z

Thank you for replying. The setup is as follow; pstn-----sip----cucm-----sip------trio-vmail I can apply the script at the outgoing invite to the trio-vmail. I still need some help to create the script. Here is what sendt in the Invite header; Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes Diversion: "nn" ;reason=unconditional;privacy=off;screen=yes If I convert the diversion headers into History-Info headers for outbound INVITE messages. And then remove the Diversion header. This is what I got; M = {} function M.outbound_INVITE(msg) if msg:getHeader("Diversion") then msg:convertDiversionToHI() msg:removeHeader("Diversion") end end return M [b]History-Info: ;index=1[/b] History-Info: ;index=1.1 History-Info: ;index=1.1.1 The Trio System understands both diversion headers and History-Info headers. If I first convert the diversion headers into History-Info headers, then I got information I need in index=1. If I now remove all other index lines everything should be ok.

RE: Delete the top-most diversion header or change order of Multiple Divers

2011-11-08T18:59:36Z

Okay, but the call flow is what? pstn-----sip----cucm-----sip?------trio-vmail Is your integration from cucm to trio a sip trunk as well? Obviously, this will determine where you apply your script. Also, the next thing you need to do is set up a place where you can make packet captures. This will give you an idea of what values you have for attempting to repopulate the dh with the original called number. For example, a way to reach back into the originally called number, may be the PAI header. At this point, you could possibly do logic similar to: -gather pai header -remove dh -add dh -repopulate dh with pai A lot will depend on your call flow. Charles

RE: Delete the top-most diversion header or change order of Multiple Divers

2011-11-08T14:14:32Z

Hi! This is an attendant system (Trio Enterprise). The system is not able to interoperate the diversion header different. I need a SIP Normalization Scripts that change the header. Support both diversion and history header.