<html><body><div style="color:#000; background-color:#fff; font-family:tahoma, new york, times, serif;font-size:12pt"><div>How does voice recorder works? Do I need RSPAN or ERSPAN? (Ki Wi)</div><div><br></div><div>as long as i know, you need to use RSPAN.<br></div><div> </div><div>GBU<br><br></div><div>Regards,</div><br><br><div>Arnold Samuel Lesar</div><div><br><br></div><div>------------------------------------------------------<br>Note: This message was created by ARNOLD SAMUEL LESAR.<br>------------------------------------------------------</div><br><div><br></div><div>--------------------------------------------------------------------------<br>--------------------------------------------------------------------------<br>PT. PELITA RELIANCE INTERNATIONAL<br>Eka Hospital BSD City, Tangerang <br>Address : CBD Lot IX, BSD City, Tangerang <br>Phone : (+62-21) 256 555 55 <br>Fax : (+62-21) 256 555 44
<br>--------------------------------------------------------------------------<br>--------------------------------------------------------------------------<br> <blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; margin-top: 5px; padding-left: 5px;"> <div style="font-family: tahoma, new york, times, serif; font-size: 12pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <font face="Arial" size="2"> <hr size="1"> <b><span style="font-weight:bold;">From:</span></b> "cisco-voip-request@puck.nether.net" <cisco-voip-request@puck.nether.net><br> <b><span style="font-weight: bold;">To:</span></b> cisco-voip@puck.nether.net <br> <b><span style="font-weight: bold;">Sent:</span></b> Wednesday, 28 December 2011, 0:00<br> <b><span style="font-weight: bold;">Subject:</span></b> cisco-voip Digest, Vol 98, Issue 16<br> </font> <br>Send cisco-voip mailing list submissions to<br>
<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>or, via email, send a message with subject or body 'help' to<br> <a ymailto="mailto:cisco-voip-request@puck.nether.net" href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a><br><br>You can reach the person managing the list at<br> <a ymailto="mailto:cisco-voip-owner@puck.nether.net" href="mailto:cisco-voip-owner@puck.nether.net">cisco-voip-owner@puck.nether.net</a><br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of cisco-voip digest..."<br><br><br>Today's Topics:<br><br> 1. Re: how do calling
party transformations work anyways?<br> (Dennis Heim)<br> 2. How does voice recorder works? Do I need RSPAN or ERSPAN? (Ki Wi)<br> 3. Re: How does voice recorder works? Do I need RSPAN or ERSPAN?<br> (Buchanan, James)<br> 4. Re: How does voice recorder works? Do I need RSPAN or ERSPAN?<br> (Dennis Heim)<br> 5. Re: How does voice recorder works? Do I need RSPAN or ERSPAN?<br> (Ki Wi)<br> 6. Re: Planning to Migrate a CUCM 7.1 to a new CUCM 8.6 Cluster<br> with Security turned on (Robert Schuknecht)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Mon, 26 Dec 2011 18:59:22 +0000<br>From: Dennis Heim <<a
ymailto="mailto:Dennis.Heim@cdw.com" href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>><br>To: Lelio Fulgenzi <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>, Mike Lydick<br> <<a ymailto="mailto:mike.lydick@gmail.com" href="mailto:mike.lydick@gmail.com">mike.lydick@gmail.com</a>><br>Cc: "<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Subject: Re: [cisco-voip] how do calling party transformations work<br> anyways?<br>Message-ID:<br> <<a ymailto="mailto:7BA674F8A3CAAF4FADB174DB8C255E8A14F146@EXMBSW2VH.corp.cdw.com"
href="mailto:7BA674F8A3CAAF4FADB174DB8C255E8A14F146@EXMBSW2VH.corp.cdw.com">7BA674F8A3CAAF4FADB174DB8C255E8A14F146@EXMBSW2VH.corp.cdw.com</a>><br>Content-Type: text/plain; charset="utf-8"<br><br>From my experience with transformation patterns, this is what I have noticed?<br><br>Calling Number Transformation on a phone ? For OnNet calls, will transform the number the calling number, but ignores mid-call events/updates such as being transferred. In the case of a call transfer the full untransformed number will show on the phone. In the case of incoming PSTN calls, this will change the way the number appears, such as stripped off the area code for local number, if users are only used to seeing 7 digits, etc. This is not the number that will appear in the directories.<br><br>Called Transformation on a phone ? IP On-net calls are not affected. If Phone A, calls, Phone B, it will always show the full extension of phone B, unless you use a
translation pattern.<br><br>Calling Number Transformation on a gateway ? Used to globalize the incoming number, such as adding a ?+? or a ?9? or whatever you want to do so the user would not need to do edit dial. For incoming calls, the transformed number would determine what shows up in the phone directory?s (missed calls, etc).<br><br>Called Number Transformation on a gateway -- ? Prior to the call being sent to the gateway, the transformation pattern will impact call routing, such as gateway localization, removing the ?+?, adding a ?1? or remove a ?1? or removing the area codes.<br><br><br>Dennis Heim<br>Senior Engineer (Unified Communications)<br>CDW Advanced Technology Services<br>10610 9th Place<br>Bellevue, WA 98004<br><br>425.310.5299 Single Number Reach (WA)<br>317.569.4255 Single Number Reach (IN)<br>317.569.4201 Fax<br><a ymailto="mailto:dennis.heim@cdw.com" href="mailto:dennis.heim@cdw.com">dennis.heim@cdw.com</a><mailto:<a
ymailto="mailto:dennis.heim@cdw.com" href="mailto:dennis.heim@cdw.com">dennis.heim@cdw.com</a>><br>cdw.com/content/solutions/unified-communications/<<a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">http://www.cdw.com/content/solutions/unified-communications/</a>><br><br>From: <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Lelio Fulgenzi<br>Sent: Friday, December 23, 2011 1:45 PM<br>To: Mike Lydick<br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>Subject: Re: [cisco-voip] how do calling party transformations work anyways?<br><br>My end goal was to be able to display the
internal extension on calls to remote destinations configured in the remote destination profile. This would allow people to know exactly which extension is calling them rather than the generic external calling mask programmed on each phone.<br><br>I'm actually testing things on a 7940, but I can probably try things on a 7942 to see if I can at least get a baseline.<br><br>I guess if this doesn't pan out, I could create a route pattern with a special filter for all remote destinations which includes uses XXXXX as the mask and see how calls would be presented, but I'm guessing that would affect off-campus calls being routed to that device as well.<br><br>Lelio<br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget
it.<br> - LFJ (with apologies to Mr. Popeil)<br><br>________________________________<br>From: "Mike Lydick" <<a ymailto="mailto:mike.lydick@gmail.com" href="mailto:mike.lydick@gmail.com">mike.lydick@gmail.com</a><mailto:<a ymailto="mailto:mike.lydick@gmail.com" href="mailto:mike.lydick@gmail.com">mike.lydick@gmail.com</a>>><br>To: "Lelio Fulgenzi" <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>>, <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Sent: Friday, December 23, 2011
12:41:27 PM<br>Subject: Re: [cisco-voip] how do calling party transformations work anyways?<br><br>So are you trying to localized the calling number? Just for the phone display. This will be applied to the phone.<br>2 setting, the latter disables or ignores the first.<br><br>Calling party Transformation CSS drop down<br>Use device Pool Calling Party Transformation<br><br>My experience is that older generation phones do not support this feature or not consistently. Spent quite a few hrs on with TAC trying to get this to work with 7941/7961/7921 and had varied results. 79x2 -79x5 this works well.<br><br>The transformation affects just the display not the routing. So in this case the call logs will show the actual number (Missed/Received calls).<br><br>There is<br>Best Regards,<br><br>Mike Lydick<br><br><br><br>On Fri, Dec 23, 2011 at 12:27 PM, Lelio Fulgenzi <<a ymailto="mailto:lelio@uoguelph.ca"
href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>> wrote:<br>Haven't tried it on the GW, not sure I want to touch those.<br><br>The fact that you can select the "external calling mask" makes me to believe that it is the devices calling party information it sends out that is affected.<br><br>ugh.<br><br>Will have to worry about this in the new year...<br><br><br><br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<br> - LFJ (with apologies to Mr. Popeil)<br><br>________________________________<br>From:
"paul dial" <<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a><mailto:<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>>><br>To: "Lelio Fulgenzi" <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>>><br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Sent: Friday, December 23, 2011 12:20:14 PM<br><br>Subject: Re: [cisco-voip] how do calling party transformations work anyways?<br><br>Agreed, very confusing. They use the word "localize" in the Help-> This Page doc for phone configuration, which makes me believe
its transforming the calling party that is displayed when someone calls this device. Does the transform work on the GW?<br><br>paul<br><br>On 12/23/2011 9:58 AM, Lelio Fulgenzi wrote:<br>I guess that's the question, does this transformation transform the calling party mask this device uses to make outbound calls, or the calling party that is displayed when someone calls this device.<br><br>So confusing.<br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<br> - LFJ (with apologies to Mr. Popeil)<br><br>________________________________<br>From: "paul dial" <<a ymailto="mailto:dialp@ucar.edu"
href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><mailto:<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><br>To: "Lelio Fulgenzi" <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Sent: Friday, December 23, 2011 11:52:28 AM<br>Subject: Re: [cisco-voip] how do calling party transformations work anyways?<br><br>I believe the Calling party transformation on the phone is used to localize the calling party number for an external call coming into your phone system. I'm guessing here, but I think you
want to localize the calling number for an internal to external call, in which case you'd want to apply it on the GW.<br><br>If the remote destination is still an IP phone under your control, then I think applying the transform to the phone would work.<br><br>paul<br><br>On 12/23/2011 9:23 AM, Lelio Fulgenzi wrote:<br>Turns out you can apply it to the device (phone), but I can't seem to get it working. Not sure if my upstream configs are overwriting it or not though.<br><br>I will have to do some more troubleshooting......ugh, I haven't looked at CallManager traces in forever.<br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<br>
- LFJ (with apologies to Mr. Popeil)<br><br>________________________________<br>From: "paul dial" <<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><mailto:<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><br>To: "Lelio Fulgenzi" <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Sent: Friday, December 23, 2011 11:20:56 AM<br>Subject: Re: [cisco-voip] how do calling party transformations work anyways?<br><br>You can apply the calling party
transformation on the Device Pool or (at least for MGCP GW) in the "Call Routing Information - Outbound Calls" section of the gateway configuration page. There might be other locations too, but I think you'd want to put it as close to the destination as possible, the idea being that if you have a different local calling number presentation standard (i.e. 7 vs 10 digits, etc) at your remote destinations, you can customize for each remote location.<br><br>paul<br><br>On 12/23/2011 8:43 AM, Lelio Fulgenzi wrote:<br>sorry, i guess the question should read, where do i apply the CSS that contains the partition that contains the transformations.<br><br><br><br>________________________________<br>From: "Lelio Fulgenzi" <<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><mailto:<a ymailto="mailto:lelio@uoguelph.ca" href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>><br>To: <a
ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Sent: Friday, December 23, 2011 10:42:22 AM<br>Subject: [cisco-voip] how do calling party transformations work anyways?<br>OK, still on this remote destination kick, trying to see how we can make things a bit better.<br><br>I'd like to be able to display the extension on the remote destination rather than the external calling mask (which is the same for everybody).<br><br>I was thinking of using a calling party transformation mask, but I can't seem to find where to apply the darn thing. If I have to create a transformation for each remote destination, I might be able to live with that, but I just wanna see it work for now.<br><br>Going to CCO now....<br><br>Any ideas in the meantime?<br><br>---<br>Lelio
Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<br> - LFJ (with apologies to Mr. Popeil)<br><br><br>_______________________________________________<br>cisco-voip mailing list<br><a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip"
target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br><br><br>_______________________________________________<br><br>cisco-voip mailing list<br><br><a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br><br><br>--<br><br>----<br><br>Paul Dial<br><br>Network Engineer<br><br>National Center for Atmospheric Research<br><br>303-497-1261<tel:303-497-1261><br><br><a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a><mailto:<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><br><br><br>--<br><br>----<br><br>Paul
Dial<br><br>Network Engineer<br><br>National Center for Atmospheric Research<br><br>303-497-1261<tel:303-497-1261><br><br><a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a><mailto:<a ymailto="mailto:dialp@ucar.edu" href="mailto:dialp@ucar.edu">dialp@ucar.edu</a>><br><br>_______________________________________________<br>cisco-voip mailing list<br><a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a
href="https://puck.nether.net/pipermail/cisco-voip/attachments/20111226/6ebbd32e/attachment-0001.html" target="_blank">https://puck.nether.net/pipermail/cisco-voip/attachments/20111226/6ebbd32e/attachment-0001.html</a>><br><br>------------------------------<br><br>Message: 2<br>Date: Tue, 27 Dec 2011 12:43:41 +0800<br>From: Ki Wi <<a ymailto="mailto:kiwi.voice@gmail.com" href="mailto:kiwi.voice@gmail.com">kiwi.voice@gmail.com</a>><br>To: Cisco VoIP List <<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Subject: [cisco-voip] How does voice recorder works? Do I need RSPAN<br> or ERSPAN?<br>Message-ID:<br> <<a ymailto="mailto:CAMefa2rDy8nn5DbjQwuAU7yo-G7HZRcoAsQ4-Y0p9csELSFq5g@mail.gmail.com"
href="mailto:CAMefa2rDy8nn5DbjQwuAU7yo-G7HZRcoAsQ4-Y0p9csELSFq5g@mail.gmail.com">CAMefa2rDy8nn5DbjQwuAU7yo-G7HZRcoAsQ4-Y0p9csELSFq5g@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>I have a setup with 2 DC . There's couple of voice gateway and cucm in each<br>DC.<br><br>There's a couple of distribution switches as well.<br><br>Between 2 DC and distribution switches are all L3.<br><br>Today, my requirement is only to span a few ports connected to the access<br>switches of distribution switch "A". If I do a rspan from those IP phones<br>back to distribution switch "A" and I will connect the voice recorder to<br>distribution switch "A", will it work?<br><br>Is it neccessary for me to span the traffic from cucm and voice gateway as<br>well? If so, ERSPAN is required.<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a
href="https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/0472a133/attachment-0001.html" target="_blank">https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/0472a133/attachment-0001.html</a>><br><br>------------------------------<br><br>Message: 3<br>Date: Tue, 27 Dec 2011 00:06:57 -0500<br>From: "Buchanan, James" <<a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>To: Ki Wi <<a ymailto="mailto:kiwi.voice@gmail.com" href="mailto:kiwi.voice@gmail.com">kiwi.voice@gmail.com</a>>, Cisco VoIP List<br> <<a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Subject: Re: [cisco-voip] How does voice recorder works? Do I need<br> RSPAN or ERSPAN?<br>Message-ID:<br> <<a
ymailto="mailto:853687641CF93E40B362F112A3D8F3CE3E87223E@SOEXCH01.Presidio.Corp" href="mailto:853687641CF93E40B362F112A3D8F3CE3E87223E@SOEXCH01.Presidio.Corp">853687641CF93E40B362F112A3D8F3CE3E87223E@SOEXCH01.Presidio.Corp</a>><br>Content-Type: text/plain; charset="us-ascii"<br><br>Depends on the voice recording product you are using. The newer ones don't require SPAN at all in many cases.<br><br>James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions<br>12 Cadillac Dr Ste 130 Brentwood, TN 37027 | <a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a><mailto:<a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>D: 615-866-5729 | F:615-866-5781 www.presidio.com<<a href="http://www.presidio.com/" target="_blank">http://www.presidio.com/</a>><br><br>From: <a
ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Ki Wi<br>Sent: Monday, December 26, 2011 10:44 PM<br>To: Cisco VoIP List<br>Subject: [cisco-voip] How does voice recorder works? Do I need RSPAN or ERSPAN?<br><br>I have a setup with 2 DC . There's couple of voice gateway and cucm in each DC.<br><br>There's a couple of distribution switches as well.<br><br>Between 2 DC and distribution switches are all L3.<br><br>Today, my requirement is only to span a few ports connected to the access switches of distribution switch "A". If I do a rspan from those IP phones back to distribution switch "A" and I will connect the voice recorder to distribution switch "A", will it work?<br><br>Is it neccessary for me to
span the traffic from cucm and voice gateway as well? If so, ERSPAN is required.<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a href="https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/4b6e0c01/attachment-0001.html" target="_blank">https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/4b6e0c01/attachment-0001.html</a>><br><br>------------------------------<br><br>Message: 4<br>Date: Tue, 27 Dec 2011 06:21:26 +0000<br>From: Dennis Heim <<a ymailto="mailto:Dennis.Heim@cdw.com" href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>><br>To: "Buchanan, James" <<a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>>, Ki Wi<br> <<a ymailto="mailto:kiwi.voice@gmail.com" href="mailto:kiwi.voice@gmail.com">kiwi.voice@gmail.com</a>>, Cisco VoIP List <<a
ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Subject: Re: [cisco-voip] How does voice recorder works? Do I need<br> RSPAN or ERSPAN?<br>Message-ID:<br> <<a ymailto="mailto:7BA674F8A3CAAF4FADB174DB8C255E8A14F24A@EXMBSW2VH.corp.cdw.com" href="mailto:7BA674F8A3CAAF4FADB174DB8C255E8A14F24A@EXMBSW2VH.corp.cdw.com">7BA674F8A3CAAF4FADB174DB8C255E8A14F24A@EXMBSW2VH.corp.cdw.com</a>><br>Content-Type: text/plain; charset="us-ascii"<br><br>Many of the products offer either one of the following or a combination of the above: (1)Port Mirroring (2) Built-In Bridge (BiB) (3) Software on PC. You also need to determine if you need to record all calls, external only, internal only, etc.<br><br>Dennis Heim<br>Senior Engineer (Unified Communications)<br>CDW Advanced Technology Services<br>10610 9th
Place<br>Bellevue, WA 98004<br><br>425.310.5299 Single Number Reach (WA)<br>317.569.4255 Single Number Reach (IN)<br>317.569.4201 Fax<br><a ymailto="mailto:dennis.heim@cdw.com" href="mailto:dennis.heim@cdw.com">dennis.heim@cdw.com</a><mailto:<a ymailto="mailto:dennis.heim@cdw.com" href="mailto:dennis.heim@cdw.com">dennis.heim@cdw.com</a>><br>cdw.com/content/solutions/unified-communications/<<a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">http://www.cdw.com/content/solutions/unified-communications/</a>><br><br>From: <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Buchanan, James<br>Sent: Tuesday, December 27, 2011 12:07 AM<br>To: Ki Wi; Cisco VoIP
List<br>Subject: Re: [cisco-voip] How does voice recorder works? Do I need RSPAN or ERSPAN?<br><br>Depends on the voice recording product you are using. The newer ones don't require SPAN at all in many cases.<br><br>James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions<br>12 Cadillac Dr Ste 130 Brentwood, TN 37027 | <a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a><mailto:<a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>D: 615-866-5729 | F:615-866-5781 www.presidio.com<<a href="http://www.presidio.com/" target="_blank">http://www.presidio.com/</a>><br><br>From: <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net"
href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>]<mailto:[mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>]> On Behalf Of Ki Wi<br>Sent: Monday, December 26, 2011 10:44 PM<br>To: Cisco VoIP List<br>Subject: [cisco-voip] How does voice recorder works? Do I need RSPAN or ERSPAN?<br><br>I have a setup with 2 DC . There's couple of voice gateway and cucm in each DC.<br><br>There's a couple of distribution switches as well.<br><br>Between 2 DC and distribution switches are all L3.<br><br>Today, my requirement is only to span a few ports connected to the access switches of distribution switch "A". If I do a rspan from those IP phones back to distribution switch
"A" and I will connect the voice recorder to distribution switch "A", will it work?<br><br>Is it neccessary for me to span the traffic from cucm and voice gateway as well? If so, ERSPAN is required.<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a href="https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/ab99fd0d/attachment-0001.html" target="_blank">https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/ab99fd0d/attachment-0001.html</a>><br><br>------------------------------<br><br>Message: 5<br>Date: Tue, 27 Dec 2011 14:56:27 +0800<br>From: Ki Wi <<a ymailto="mailto:kiwi.voice@gmail.com" href="mailto:kiwi.voice@gmail.com">kiwi.voice@gmail.com</a>><br>To: Dennis Heim <<a ymailto="mailto:Dennis.Heim@cdw.com" href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>><br>Cc: Cisco VoIP List <<a ymailto="mailto:cisco-voip@puck.nether.net"
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>Subject: Re: [cisco-voip] How does voice recorder works? Do I need<br> RSPAN or ERSPAN?<br>Message-ID:<br> <<a ymailto="mailto:CAMefa2rrVsKfZydm-jTeKvArjUdrc3doCD4QxZcmGg4O_c7g6g@mail.gmail.com" href="mailto:CAMefa2rrVsKfZydm-jTeKvArjUdrc3doCD4QxZcmGg4O_c7g6g@mail.gmail.com">CAMefa2rrVsKfZydm-jTeKvArjUdrc3doCD4QxZcmGg4O_c7g6g@mail.gmail.com</a>><br>Content-Type: text/plain; charset="windows-1252"<br><br>Hi Dennis/James,<br>Thanks for the quick reply.<br><br>The product that the customer have actually works for port mirror<br>scenarios. I'm aware that those products are actually looking for SCCP<br>messages during their capture. However, I'm wondering is the SCCP messages<br>from the phone port (will see both tx/rx packets to the phone) sufficient ?<br>Or do I need to span extra signalling messages from CUCM and as
well as<br>those extra rtp stream from voice gateway?<br><br>Currently, my best practise is i will span all the rx traffic into the<br>voice vlan (ingress traffic into switchport) and rspan it to the recorder.<br>However, now i'm hoping that I can simply sniff both tx/rx packets on the<br>few IP phones that requires recording.<br><br>*imaging the site have 2000 phones and i only need to record 5 ip phones<br>conversation in the network, the software vendor is worried about the cpu<br>utilization as well if we were to enable span traffic everywhere*<br><br><br><br>On Tue, Dec 27, 2011 at 2:21 PM, Dennis Heim <<a ymailto="mailto:Dennis.Heim@cdw.com" href="mailto:Dennis.Heim@cdw.com">Dennis.Heim@cdw.com</a>> wrote:<br><br>> Many of the products offer either one of the following or a combination<br>> of the above: (1)Port Mirroring (2) Built-In Bridge (BiB) (3) Software on<br>> PC. You also need to determine if you need to record all
calls, external<br>> only, internal only, etc.****<br>><br>> ** **<br>><br>> Dennis Heim<br>> Senior Engineer (Unified Communications)<br>> CDW Advanced Technology Services<br>> 10610 9th Place<br>> Bellevue, WA 98004<br>><br>> 425.310.5299 Single Number Reach (WA)****<br>><br>> 317.569.4255 Single Number Reach (IN)<br>> 317.569.4201 Fax<br>> <a ymailto="mailto:dennis.heim@cdw.com" href="mailto:dennis.heim@cdw.com">dennis.heim@cdw.com</a>*<br>> *cdw.com/content/solutions/unified-communications/<<a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">http://www.cdw.com/content/solutions/unified-communications/</a>><br>> ****<br>><br>> ** **<br>><br>> *From:* <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<br>> <a
ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] *On Behalf Of *Buchanan, James<br>> *Sent:* Tuesday, December 27, 2011 12:07 AM<br>> *To:* Ki Wi; Cisco VoIP List<br>> *Subject:* Re: [cisco-voip] How does voice recorder works? Do I need<br>> RSPAN or ERSPAN?****<br>><br>> ** **<br>><br>> Depends on the voice recording product you are using. The newer ones don?t<br>> require SPAN at all in many cases.****<br>><br>> ** **<br>><br>> *James Buchanan*|* UC Technology Manager *| *Presidio South *|*Presidio Networked Solutions<br>> *****<br>><br>> *12 Cadillac Dr Ste 130 Brentwood, TN 37027 *|* **<a ymailto="mailto:jbuchanan@presidio.com" href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>**<br>> *<br>><br>> *D: 615-866-5729* | *F:615-866-5781* *www.presidio.com*<<a
href="http://www.presidio.com/" target="_blank">http://www.presidio.com/</a>><br>> ****<br>><br>> ** **<br>><br>> *From:* <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] *On Behalf Of *Ki Wi<br>> *Sent:* Monday, December 26, 2011 10:44 PM<br>> *To:* Cisco VoIP List<br>> *Subject:* [cisco-voip] How does voice recorder works? Do I need RSPAN or<br>> ERSPAN?****<br>><br>> ** **<br>><br>> I have a setup with 2 DC . There's couple of voice gateway and cucm in<br>> each DC.<br>><br>> There's a couple of distribution switches as well.<br>><br>> Between 2 DC and distribution switches are all L3.<br>><br>> Today, my requirement is only to span a few
ports connected to the access<br>> switches of distribution switch "A". If I do a rspan from those IP phones<br>> back to distribution switch "A" and I will connect the voice recorder to<br>> distribution switch "A", will it work?<br>><br>> Is it neccessary for me to span the traffic from cucm and voice gateway as<br>> well? If so, ERSPAN is required. ****<br>><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a href="https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/6a7d1283/attachment-0001.html" target="_blank">https://puck.nether.net/pipermail/cisco-voip/attachments/20111227/6a7d1283/attachment-0001.html</a>><br><br>------------------------------<br><br>Message: 6<br>Date: Tue, 27 Dec 2011 13:28:24 +0100<br>From: "Robert Schuknecht" <<a ymailto="mailto:rschuknecht@gmx.de" href="mailto:rschuknecht@gmx.de">rschuknecht@gmx.de</a>><br>To: "'Wes Sisk'" <<a
ymailto="mailto:wsisk@cisco.com" href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>><br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>Subject: Re: [cisco-voip] Planning to Migrate a CUCM 7.1 to a new CUCM<br> 8.6 Cluster with Security turned on<br>Message-ID: <01e901ccc493$07781a90$16684fb0$@gmx.de><br>Content-Type: text/plain; charset="utf-8"<br><br>Hi List,<br><br>It?s a long time since the original post but, finally we got it working, at least in our LAB. <br><br>Here are the steps we used to get it working:<br><br>1) Included the new 8.6 TFTP Server in the CTL File of the old cluster<br>2) Configured the Phones on the new Cluster with the option "Upgrade/Install new CTL by Null String / by MIC"<br>3) Deleted the Phones or changed their MAC on the Old Cluster, so the Phones are no longer known by the old
Cluster<br>4) Rebooted the Phones on the old Cluster<br>5) Phones got their configuration Files from the new Cluster (old Cluster TFTP did not know about the Phones so he queried the TFTP of the new Cluster)<br>6) Phones got the CTL File of the new Cluster<br>7) Phones registered successfully to the new Cluster <br><br>To all who have responded, Thanks for your help!<br><br>/Robert<br><br>-----Urspr?ngliche Nachricht-----<br>Von: <a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a ymailto="mailto:cisco-voip-bounces@puck.nether.net" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] Im Auftrag von <a ymailto="mailto:rschuknecht@gmx.de" href="mailto:rschuknecht@gmx.de">rschuknecht@gmx.de</a><br>Gesendet: Tuesday, November 08, 2011 12:53 PM<br>An: Wes Sisk<br>Cc: <a ymailto="mailto:cisco-voip@puck.nether.net"
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>Betreff: Re: [cisco-voip] Planning to Migrate a CUCM 7.1 to a new CUCM 8.6 Cluster with Security turned on<br><br>Wes,<br><br>thanks for info. We will give it another try. I will let you how it went.<br><br>/Robert<br>-------- Original-Nachricht --------<br>> Datum: Mon, 7 Nov 2011 18:20:58 -0500<br>> Von: Wes Sisk <<a ymailto="mailto:wsisk@cisco.com" href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>><br>> An: <a ymailto="mailto:rschuknecht@gmx.de" href="mailto:rschuknecht@gmx.de">rschuknecht@gmx.de</a><br>> CC: <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>> Betreff: Re: [cisco-voip] Planning to Migrate a CUCM 7.1 to a new CUCM <br>> 8.6 Cluster with Security turned on<br><br>> A few things stand out here. Hopefully Jason will chime in with <br>> whatever I miss on the
security aspects.<br>> * Centralized TFTP requires using the *newest* version as the centralized<br>> TFTP: CSCsq48448 CCM SRND centralized TFTP needs to require central TFTP<br>> be highest ver<br>> <br>> I believe we summarized the process this way:<br>> <br>> pre 8.0 security with hardware etoken<br>> ? update old cluster CTL to include new cluster TFTP server. sign <br>> old CTL with etoken<br>> ? should not be necessary if new cluster CTL file is signed by the <br>> same eToken<br>> ? Late (9.x?) TNP phones will trust new TFTP because signed with <br>> same hardware etoken(unverified)<br>> ? Older phone loads require updating CTL to include new TFTP server<br>>
? restart TFTP old cluster to reload CTL file into cache<br>> ? reset phones on old cluster and verify CTL download<br>> ? use same hardware etoken on new cluster. just sign CTL file with <br>> same eToken. enabling security is not required.<br>> ? for pre-8.x to 8.x just use same eToken to sign CTL on 8.x cluster<br>> ? is there a way to accomplish this without generating CTL file? No.<br>> CTL files are 'permanent'. would require touching every phone to <br>> delete CTLs. No "migration" to ITL's<br>> ? if CTL file in old cluster then *must* generate CTL in new cluster<br>> ? point phones to new
cluster<br>> <br>> Regards,<br>> Wes<br>> <br>> <br>> On Nov 7, 2011, at 7:39 AM, <a ymailto="mailto:rschuknecht@gmx.de" href="mailto:rschuknecht@gmx.de">rschuknecht@gmx.de</a> wrote:<br>> <br>> Hi List,<br>> <br>> I am currently planning a CUCM Migration from Version 7.1.5 to Version <br>> 8.6, on new Hardware. The old cluster has security turned on (CTL <br>> Client an E-Token). What makes more difficult is, we have to use the <br>> old Publisher as an centralized TFTP Server for both, the old 7.1.5 and the new 8.6 Cluster.<br>> <br>> I thought it should be possible by doing it this way:<br>> <br>> - Define the 8.6 TFTP-Server as alternate TFTP on the old Server<br>> - Run the CTL-Client on the old Cluster and put the new TFTP Server in <br>> the CTL File<br>> - Run the CTL-CLient on the new Cluster and add the old TFTP Server<br>> - Configure the new Cluster with all Phones<br>> -
Delete the Phones on the old Cluster (Change MAC Add.)<br>> <br>> But in this scenario the Phones would not register to the new Cluster? <br>> <br>> Now i am looking for a strategy to configure it correctly. Any <br>> information and help is more than welcome.<br>> <br>> /Robert<br>> -- <br>> NEU: FreePhone - 0ct/min Handyspartarif mit Geld-zur?ck-Garantie! <br>> Jetzt informieren: <a href="http://www.gmx.net/de/go/freephone" target="_blank">http://www.gmx.net/de/go/freephone</a><br>> _______________________________________________<br>> cisco-voip mailing list<br>> <a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>> <br>> <br><br>--<br>Empfehlen Sie GMX DSL Ihren
Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! <a href="https://freundschaftswerbung.gmx.de" target="_blank">https://freundschaftswerbung.gmx.de</a><br>_______________________________________________<br>cisco-voip mailing list<br><a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br><br><br><br>------------------------------<br><br>_______________________________________________<br>cisco-voip mailing list<br><a ymailto="mailto:cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br><br>End of cisco-voip Digest, Vol 98, Issue
16<br>******************************************<br><br><br> </div> </div> </blockquote> </div></div></body></html>