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- About Gaurav Purohit
07-18-2018
Gaurav Purohit
Member since
12-09-2010
Beginner
Created by
Gaurav Purohit
in Collaboration, Voice and Video Documents
in Collaboration, Voice and Video Documents
04-18-2018
06:51 PM
04-18-2018
06:51 PM
Hi I was recently having hard time to get RTMT to work. Here are some tips I figured out:
1) I was trying to install RTMT 11.5 which wouldn't install until I manually installed the latest java version (I had to use older versions as some clunky old applications only work on java6).
2) After successful installation, Everytime I try to run RTMT (even as Admin) nothing would happen. This was really frustrating as there was nothing in the error logs as well. I therefore started to look at the install log and realised RTMT installs its own jre which to my suspicion was not consistent with the version I manually installed. so I replaced the jre in "C:\Program Files (x86)\Cisco\Unified Rtmt\JRtmt" with the jre from actual java install dir "C:\Program Files (x86)\Java" (for CUCM11.5 this was 1.8).
After replacing the jre with the installed/system jre, the RTMT would fire up and authenticate but it'd fail with error "RTMT failed to initialize. Exiting".
3) I ran the bat file where RTMT is installed, this gave the indication where things were failing
"C:\Program Files (x86)\Cisco\Unified Rtmt\JRtmt\run.bat"
In my case, it was failing to find "ZI" folder in the newly replaced jre. It seems this folder has got something to do with timezones which RTMT is paranoid about.
After I copied the "ZI" folder and all its content from default RTMT installed jre, to the new lib folder under install directory("C:\Program Files (x86)\Cisco\Unified Rtmt\JRtmt\lib\"), everything started to work.
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Created by
Gaurav Purohit
in Contact Center
in Contact Center
11-05-2017
06:31 PM
11-05-2017
06:31 PM
seems like an issue with the audio file itself. can you test any other file? does it happen with any other audio file?
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Created by
Gaurav Purohit
in Call Control
in Call Control
04-04-2017
07:56 PM
04-04-2017
07:56 PM
Thanks heaps! Can't believe I was so close.
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Created by
Gaurav Purohit
in Call Control
in Call Control
04-03-2017
06:19 PM
04-03-2017
06:19 PM
Hi, I'm struggling to get my LUA script to work. The RightFax server on other end does not allow SIP parameters to be altered and sends a basic invite with no privacy option. I'd like to alter my inbound invites to add P-Asserted-ID and alter From field. Here is my LUA script: M = {} trace.enable() function M.inbound_INVITE(msg) --get Header Values local uriString = msg:getUri("From") local nuri = sipUtils.parseUri(uriString) trace.format("Parse success") local nUser = nuri:getUser() trace.format("user is '%s'", nUser) local nHost = nuri:getHost() trace.format("host is '%s'", nHost) --get tag from From header local fromtag = msg:getHeaderValueParameter("From", "tag") trace.format("tag from From is '%s'", fromtag) --create new from header local newFromHeader = string.format("\"Anonymous\" <sip:Anonymous@%s>;tag=%s\r\n", nHost, fromtag) trace.format("new From Header is '%s'", newFromHeader) msg:modifyHeader("From", newFromHeader) trace.format("From Header is updated successfully") --create Remote party ID string with privacy local newRPartyHeader = string.format("<sip:%s@%s>;party=calling;id-type=subscriber;privacy=full;screen=yes\r\n", nUser, nHost) trace.format("header string created '%s'", newRPartyHeader) --Add Remote party ID (not added for now) --msg:addHeader("Remote-Party-ID", newRPartyHeader) local newPAIHeader = string.format("\"RightFax\" <sip:%s@%s>\r\n", nUser, nHost) trace.format("PAI header string created '%s'", newPAIHeader) msg:addHeader("P-Asserted-Identity", newPAIHeader) local ids = string.format("id\r\n") trace.format("Adding ID") msg:addHeader("Privacy", ids) trace.format("All Done\r\n") end return M Original Invite from RightFax: INVITE sip:0385410508@172.22.5.31 SIP/2.0 From: Rightfax Dev <sip:0397277670@172.22.31.185>;tag=8be2690-b91f16ac-13c4-55013-312cf8-26efef6b-312cf8 To: <sip:0385410508@172.22.5.31> Call-ID: 776a928-b91f16ac-13c4-55013-312cf8-70e1aefd-312cf8 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.22.31.185:5060;branch=z9hG4bK-312cf8-c017a9f1-27c0c9e1 Supported: 100rel Max-Forwards: 70 User-Agent: Brktsip/6.5.2B5 (Dialogic) Contact: <sip:172.22.31.185:5060> Content-Type: application/sdp Content-Length: 150 v=0 o=- 2212556890 0340060000 IN IP4 172.22.31.185 s=no_session_name t=0 0 m=audio 56300 RTP/AVP 0 c=IN IP4 172.22.31.185 a=rtpmap:0 pcmu/8000 Altered Invite from RightFax: INVITE sip:0385410508@172.22.5.31 SIP/2.0 From: "Anonymous" <sip:Anonymous@172.22.31.185>;tag=8be2690-b91f16ac-13c4-55013-312cf8-26efef6b-312cf8 P-Asserted-Identity: "RightFax" <sip:0397277670@172.22.31.185> Supported: 100rel Content-Length: 150 User-Agent: Brktsip/6.5.2B5 (Dialogic) Privacy: id To: <sip:0385410508@172.22.5.31> Contact: <sip:172.22.31.185:5060> Content-Type: application/sdp Call-ID: 776a928-b91f16ac-13c4-55013-312cf8-70e1aefd-312cf8 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.22.31.185:5060;branch=z9hG4bK-312cf8-c017a9f1-27c0c9e1 Max-Forwards: 70 v=0 o=- 2212556890 0340060000 IN IP4 172.22.31.185 s=no_session_name t=0 0 m=audio 56300 RTP/AVP 0 c=IN IP4 172.22.31.185 a=rtpmap:0 pcmu/8000 Error: SIP/2.0 400 Bad Request - 'Malformed/Missing FROM: field'
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Labels:
Created by
Gaurav Purohit
in Contact Center
in Contact Center
01-18-2017
07:08 PM
01-18-2017
07:08 PM
Hi, what you want is very simple and no need for a database. You don't even need a digit menu element, you can create a Call studio application for CVP with an audio saying something like "do you need to Close the Contact Centre" within a "yes/no" element. you can catch this response and send it back to ICM which can then do the decision. The variable is automagically created on CVP side once passed from ICM while invoking the CVP application using To VXML Array. To pass the data back, CVP uses Caller Input or one of the From ExtVXML Array elements. user and global variables will not help when you want external application to work for you (CVP being external to ICM).
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Created by
Gaurav Purohit
in Management
in Management
06-30-2016
10:13 PM
06-30-2016
10:13 PM
Hi, just wondering if you got it to work? I'm getting same error. Thanks
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Created by
Gaurav Purohit
in Contact Center
in Contact Center
02-10-2016
04:02 PM
02-10-2016
04:02 PM
Hi Faisal, you can specify the Media server while invoking the CVP Application. this will give you some customization capability as you can specify the one at ICM.
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01-28-2016
05:03 PM
Hi David,
have a look at this great writeup:
https://supportforums.cisco.com/document/63806/using-termination-call-detail-and-route-call-detail-records-troubleshooting
Also, you will not be able to capture the Agent ID to a PV as it happens on a different leg and a different GUID gets assigned to that leg. The only way to link the two legs is via RouterCallKey. It can take upto 30 minutes or more(or less) for this leg's entry to appear in TCD table (the leg where AgentPeripheralNumber can be seen).
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12-22-2015
08:18 PM
Hi,
you need to collect call manager detail traces (and perhaps packet captures through CLI). some pointers:
1) you have "use MTP if needed", which means if they send SDP in invite (which they must be with G.711 only), you are going to forward it as it is.
2) whats the MOH type? if Multicast, is it allowed?
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Created by
Gaurav Purohit
in IP Telephony and Phones
in IP Telephony and Phones
11-18-2015
03:52 PM
11-18-2015
03:52 PM
Also, on the 2911 make sure you do allow connection sip to sip
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11-18-2015
03:45 PM
Hi,
- for early offer, please make sure annunciator is also a part of the Media Resource for your SIP trunk.
- for delayed offer, probabaly a transcoding resource is getting invoked which does't has connectivity to both end points. this is just a guess based on limited information. you really need to look into the CUCM logs to see details.
Does the consultative transfer work?
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Created by
Gaurav Purohit
in Collaboration, Voice and Video Videos
in Collaboration, Voice and Video Videos
03-31-2015
04:57 PM
03-31-2015
04:57 PM
Hi, just wondering, if there is a way to upgrade CUCM 8.6 to 10.5, along with downgrade the provisioning from 2500 users to 1000 users? is it just a matter of using the lower spec OVA for target VM? also, how would the licensing work?
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Created by
Gaurav Purohit
in IP Telephony and Phones
in IP Telephony and Phones
03-31-2015
03:03 PM
03-31-2015
03:03 PM
Hi, without looking at the logs and the topology all i can say is: 1) make sure no DSP is invoked in the 79XX's case. 2) if it is invoked, make sure there is connectivity from DSP to both the end points. firewall might be blocking the RTP across. 3) capture the logs from CUCM and see what are the two parties trying to negotiate at. HTH
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Created by
Gaurav Purohit
in Collaboration, Voice and Video Documents
04-18-2018 06:51 PM
04-18-2018 06:51 PM
Hi I was recently having hard time to get RTMT to work. Here are some
tips I figured out: 1) I was t...
11-05-2017 06:31 PM
seems like an issue with the audio file itself. can you test any other
file? does it happen with any...
Created by
Gaurav Purohit
in Call Control
04-04-2017 07:56 PM
04-04-2017 07:56 PM
Thanks heaps!Can't believe I was so close.
Created by
Gaurav Purohit
in Call Control
04-03-2017 06:19 PM
04-03-2017 06:19 PM
Hi,I'm struggling to get my LUA script to work. The RightFax server on
other end does not allow SIP ...
Created by
Gaurav Purohit
in Contact Center
01-18-2017 07:08 PM
01-18-2017 07:08 PM
Hi,what you want is very simple and no need for a database.You don't
even need a digit menu element,...
Created by
Gaurav Purohit
in Management
06-30-2016 10:13 PM
06-30-2016 10:13 PM
Hi,just wondering if you got it to work?I'm getting same error.Thanks
Created by
Gaurav Purohit
in Contact Center
02-10-2016 04:02 PM
02-10-2016 04:02 PM
Hi Faisal,you can specify the Media server while invoking the CVP
Application. this will give you so...
01-28-2016 05:03 PM
Hi David, have a look at this great writeup:
https://supportforums.cisco.com/document/63806/using-te...
12-22-2015 08:18 PM
Hi, you need to collect call manager detail traces (and perhaps packet
captures through CLI). some p...
Created by
Gaurav Purohit
in IP Telephony and Phones
11-18-2015 03:52 PM
11-18-2015 03:52 PM
Also, on the 2911 make sure you do allow connection sip to sip
11-18-2015 03:45 PM
Hi, - for early offer, please make sure annunciator is also a part of
the Media Resource for your SI...
Created by
Gaurav Purohit
in Collaboration, Voice and Video Videos
03-31-2015 04:57 PM
03-31-2015 04:57 PM
Hi,just wondering, if there is a way to upgrade CUCM 8.6 to 10.5, along
with downgrade the provision...
Public Statistics
| Date Registered | 12-09-2010 06:56 PM |
| Date Last Visited | 07-18-2018 12:03 AM |
| Total Messages Posted | 66 |
| Total Helpful Votes Received | 6 |
