<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Thanks Nick!<span><br><br><span name="x"></span>---<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><span name="x"></span><br></span><br><hr id="zwchr"><b>From: </b>"Nick Matthews" <matthnick@gmail.com><br><b>To: </b>"Wes Sisk" <wsisk@cisco.com><br><b>Cc: </b>"Lelio Fulgenzi" <lelio@uoguelph.ca>, "Nate VanMaren" <VanMarenNP@ldschurch.org>, "cisco-voip@puck.nether.net VOIP" <cisco-voip@puck.nether.net><br><b>Sent: </b>Monday, March 12, 2012 1:06:45 PM<br><b>Subject: </b>Re: [cisco-voip] H323 Gateway needed for SNR mid-call features?<br><br>Digital voice is basically PRI channels. Ex: the 2901 can fit 3-256<br>PVDM but doesn't have the horsepower to manage that many channels, so<br>it has a 100 digital voice channel capacity.<br><br>The metric you're going to be more interested in is VXML capacity.<br>It's pretty close to the CUBE and MTP numbers if you have those. They<br>aren't published since they're benchmarks and not promises. You can<br>contact your account team to get more details if you're not already<br>privy to that information.<br><br>-nick<br><br>On Mon, Mar 12, 2012 at 12:21 PM, Wes Sisk <wsisk@cisco.com> wrote:<br>> Apologies Lelio, but I honestly don't know. Maybe Nick can translate this<br>> marketing speak.<br>><br>> On Mar 9, 2012, at 3:39 PM, Lelio Fulgenzi wrote:<br>><br>> If I wanted to spec a couple of new routers for this purpose for<br>> active/active, active/standby, active/shelf (or whatever is available), what<br>> would I be looking at?<br>><br>> Is one of these H323 hairpin calls considered a "digital voice" call as<br>> spec'ed here:<br>><br>> http://www.cisco.com/en/US/prod/collateral/routers/ps10538/aag_c45_556315.pdf<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>> ________________________________<br>> From: "Wes Sisk" <wsisk@cisco.com><br>> To: "Lelio Fulgenzi" <lelio@uoguelph.ca><br>> Cc: "Nate VanMaren" <VanMarenNP@ldschurch.org>,<br>> "cisco-voip@puck.nether.net VOIP" <cisco-voip@puck.nether.net><br>> Sent: Friday, March 9, 2012 11:49:55 AM<br>> Subject: Re: [cisco-voip] H323 Gateway needed for SNR mid-call features?<br>><br>> It's been tried and it leads to interesting scenarios. invoking features<br>> like transfer utilize the h323/sip router's dialplan rather than CUCM's.<br>> Then you configure peers to route to CUCM. IF you're tandeming through<br>> CUCM to get the h.323 gateway then you tandem back to cucm so you need<br>> ip-ip-gw functionality. then you have issues with SNR dial-peers<br>> overlapping/conflicting with SRST dial-peers. That leads to some<br>> interesting COR configuration. Down the rabbit hole.<br>><br>> It was a fun lab and whiteboard session with several CCIE's. It could be<br>> made to work with numerous limitations. it's possible but may not be<br>> recommended for the masses without some very careful design and<br>> functionality choices.<br>><br>> /wes<br>><br>> On Mar 9, 2012, at 11:31 AM, Lelio Fulgenzi wrote:<br>><br>> wonder if i could run the hairpinning on the same router that has the MGCP<br>> gateways. that would probably screw up any chance of getting SRST working<br>> without confusion though.<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>> From: "Nate VanMaren" <VanMarenNP@ldschurch.org><br>> To: "Wes Sisk" <wsisk@cisco.com>, "Lelio Fulgenzi" <lelio@uoguelph.ca><br>> Cc: "cisco-voip@puck.nether.net VOIP" <cisco-voip@puck.nether.net><br>> Sent: Friday, March 9, 2012 11:02:23 AM<br>> Subject: RE: [cisco-voip] H323 Gateway needed for SNR mid-call features?<br>><br>> I think SIP is ok too. Basically no MGCP because it can’t do the tcl/vxml.<br>><br>><br>><br>> From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] On<br>> Behalf Of Wes Sisk<br>> Sent: Friday, March 09, 2012 8:50 AM<br>> To: Lelio Fulgenzi<br>> Cc: cisco-voip@puck.nether.net VOIP<br>> Subject: Re: [cisco-voip] H323 Gateway needed for SNR mid-call features?<br>><br>><br>><br>> my vague recollection is yes. mid call features depend on the h323 gateway<br>> listening in on the call and invoking a tcl or vxml script (increasing<br>> fuzziness here).<br>><br>><br>><br>> On Mar 9, 2012, at 10:00 AM, Lelio Fulgenzi wrote:<br>><br>><br>> OK, I'm pretty sure this is the case, but I'm just wondering if someone can<br>> confirm/deny.<br>><br>> Do I need an H323 gateway for SNR mid-call features?<br>><br>> The document talks about using H323 for mobile voice access, but I'm just<br>> wondering if this is a per-requisite for mid-call features like transfer,<br>> etc.<br>><br>> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmobmgr.pdf<br>><br>> Thanks, Lelio<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>> cisco-voip mailing list<br>> cisco-voip@puck.nether.net<br>> https://puck.nether.net/mailman/listinfo/cisco-voip<br>><br>><br>><br>><br>><br>> NOTICE: This email message is for the sole use of the intended recipient(s)<br>> and may contain confidential and privileged information. Any unauthorized<br>> review, use, disclosure or distribution is prohibited. If you are not the<br>> intended recipient, please contact the sender by reply email and destroy all<br>> copies of the original message.<br>><br>><br>><br>><br>><br></div></body></html>