CUCM - Nortel CS1000 SIP GW Integration Modify INVITE header

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

Integration with 3rd Party Presence service

2013-05-10T11:32:22Z

Is it possible to integrate CUCM with 3rd party presence server (not CUPS)? For example I wanted to integrate CUCM with IBM presence service or Oracle Communication Presence using SIP trunking. Do you see any major challenges in doing so? The required use cases are as follows: - CUCM should be able to publish presence of SIP clients - CUCM should be able to subscribe for buddy list both internal and external to CUCM - CUCM notfiies presence update upong receiving NOTIFY from Presence server - Presence server maintains registered buddylist. CUCM provides buddy list through register

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.

CUCM integration with Glassfish communication server

2013-01-02T07:38:16Z

Hi, Is it possible to integrate Glassfish Communication server with Cisco CUCM? With Thanks and Regards, Huthesh

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

RE: Cisco Call Recording Interface random error Reason: Q.850;cause=47

2012-11-26T21:40:15Z

This kind of error is usually seen when a transcoder, MTP or similar media-path helper resource is either not configured or has run out of available instances. Ensuring that DTMF signaling type and codec usage is unified and configured correctly may help reduce the need for such resources.

Cisco Call Recording Interface random error Reason: Q.850;cause=47

2012-11-26T20:51:38Z

Hello, One of our customers is having problem with Cisco Call Recording interface. Most of the calls getting recorded fine except in case of consulation transfer. In case of consulation transfer some calls (to the same extension) get recorded, other calls getting error Reason: Q.850;cause=47. Any suggestions, ideas? Thanks bad call Wireshark Message Header Via: SIP/2.0/UDP 10.32.60.12:5060;branch=z9hG4bK152da74fee0f0 From: "Rob Trefz Agent" ;tag=ae0b7800-6bc8-4225-972e-582b03635c76-44193253 To: ;tag=002f1860 Date: Mon, 19 Nov 2012 15:03:11 GMT Call-ID: [url=mailto:3acde700-aa14a2f-c1cb-c3c200a@10.32.60.12]3acde700-aa14a2f-c1cb-c3c200a@10.32.60.12[/url] User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 P-Preferred-Identity: "Rob Trefz Agent" CSeq: 102 BYE Reason: Q.850;cause=47 Content-Length: 0 Good Call Wireshark Message Header Via: SIP/2.0/UDP 10.32.60.12:5060;branch=z9hG4bK152b64154c8fe From: "Rob Trefz Agent" ;tag=ae0b7800-6bc8-4225-972e-582b03635c76-44193217 To: ;tag=302d2e62 Date: Mon, 19 Nov 2012 14:56:44 GMT Call-ID: [url=mailto:54226380-aa148ac-c1c1-c3c200a@10.32.60.12]54226380-aa148ac-c1c1-c3c200a@10.32.60.12[/url] User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 P-Preferred-Identity: "Rob Trefz Agent" CSeq: 102 BYE Content-Length: 0

Automatic reply: New Message from Sanjeev Thallikar in Cisco Unified Commun

2012-11-07T15:13:48Z

Am currently booked on a special assignment offsite untill Nov 15th. Expect some delays in responding to emails. Best Regards, Mohamed Lubbad

RE: SIP Dialer and Call Progress Analysis

2012-11-07T15:12:48Z

[quote=Huthesha K]Hi, I am still trying to identify a solution. Can anyone please help me in this regards? Thanks, Huthesh [/quote] Old thread but thought this will be useful for somebody in the future http://www.cisco.com/image/gif/paws/111980/cpa-00.pdf

How to monitor - alert SIP trunk state in UCM 8.6 and later

2012-10-16T21:00:41Z

Enabled SIP profile, Options Ping on the trunk. It works fine but I can't find this in RTMT to report on.. Anybody got this working? Thanks Kees

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.

RE: Cisco CME and NET VX900 SIP trunk

2012-09-28T20:22:17Z

[quote=AAD VAN DER ZANDEN]Has anyone once achieved thsi? We are trying to connect the CME SIP trunk ( 12.4.22T advipservices ) on a 2811 to a VX900 We can call from SIP phone registered on VX to phone registered on cisco CME but not back so it looks like a dial-peer issue though I can't think of anything . There is no dial tone on SIP phone at CME ( SPA922 ) Any suggestions? Here is our config: voice service voip allow-connections sip to sip sip registrar server expires max 600 min 60 ! ! voice class codec 1 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 g711alaw ! voice register global mode cme source-address xx.xx.xx.xx port 5060 max-dn 8 max-pool 8 authenticate register dialplan-pattern 2 690.... extension-length 4 tftp-path flash: create profile sync 0007272453516527 ! voice register dn 1 number 6902222 allow watch ! voice register pool 1 id mac [mac-address of client] type 7970 dtmf-relay sip-notify voice-class codec 1 username xxxx password xxxxx ! dial-peer voice 1 voip description ** SIP TRUNK ** destination-pattern 686.... voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:[IP of remote VX SIP server] dtmf-relay rtp-nte no vad ! telephony-service em logout 0:0 0:0 0:0 max-ephones 10 max-dn 20 ip source-address xx.xx.xx.xx port 2000 load 7960-7940 P00308000500 dialplan-pattern 1 690.... extension-length 4 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! [/quote] I had a problem like this also. My problem was the VX900. The CME IP source address was a loopback (192.168.20.1) and the CME router was the default gateway (192.168.1.1) of the network the VX was on. I could call from a PBX switch through the VX to the CME, but I couldn't call back. The problem was the VX was seeing the default gateway as the inbound calling party not the loopback address. To fix this I made the SIP inbound the default gaetway and everything worked fine. I guess you could also just make the CME IP source address the default gateway. I hope this helps someone, because this is the only place I could find anything about making a VX and a CME communicate. If you go to the CLI of the VX Enable->Trace all (Trace none, to turn off) and then call you can see what the issue is.

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