Re: [cisco-voip] HELP
On Thu, Nov 1, 2018 at 11:03 AM Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > And my axe! > > On Thu, Nov 1, 2018 at 8:37 AM Ryan Huff wrote: > >> In general? Or with something specific? Aside from Cisco Voice, I’m a >> pretty good carpenter and a decent auto mechanic. >> >> Sent from my iPhone >> >> On Nov 1, 2018, at 09:28, Fry, John wrote: >> >> help >> >> >> >> >> State of Illinois - CONFIDENTIALITY NOTICE: The information contained in >> this communication is confidential, may be attorney-client privileged or >> attorney work product, may constitute inside information or internal >> deliberative staff communication, and is intended only for the use of the >> addressee. Unauthorized use, disclosure or copying of this communication or >> any part thereof is strictly prohibited and may be unlawful. If you have >> received this communication in error, please notify the sender immediately >> by return e-mail and destroy this communication and all copies thereof, >> including all attachments. Receipt by an unintended recipient does not >> waive attorney-client privilege, attorney work product privilege, or any >> other exemption from disclosure. >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> >> https://nam05.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7C57316feacdef43d550a108d63ffddc40%7C84df9e7fe9f640afb435%7C1%7C0%7C636766756879501818&sdata=5cno5Xbb84KJGmeHHCyaplEtFRdpNj6xrVM%2FXiENBSc%3D&reserved=0 >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Basic corlist question
Thanks Guys, answer was staring me in the face the whole time. On Tue, Oct 30, 2018 at 6:10 PM NateCCIE wrote: > Yes. If you have no cor applied to the outbound dial peer, then anything > can use that dialpeer. > > Sent from my iPhone > > > On Oct 30, 2018, at 1:11 PM, Ed Leatherman > wrote: > > > > At least - I think this is basic... > > > > Can a call with an incoming corlist (based on the inbound dialpeer) > match an outbound dialpeer that has no corlist? Basically, following the > lock/key idea, the outbound dial-peer has no lock so anything can match it. > > > > I'm trying to go through a vendor-provided IOS config for a SIP > integration between a Cisco router and our public safety's dispatch system, > and it keeps matching on a dialpeer we are not expecting, based on how we > _think_ it is supposed to work. Essentially the scenario above. > > > > Thanks! > > > > -- > > Ed Leatherman > > ___ > > cisco-voip mailing list > > cisco-voip@puck.nether.net > > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Basic corlist question
At least - I think this is basic... Can a call with an incoming corlist (based on the inbound dialpeer) match an outbound dialpeer that has no corlist? Basically, following the lock/key idea, the outbound dial-peer has no lock so anything can match it. I'm trying to go through a vendor-provided IOS config for a SIP integration between a Cisco router and our public safety's dispatch system, and it keeps matching on a dialpeer we are not expecting, based on how we _think_ it is supposed to work. Essentially the scenario above. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] OT: Amazon Connect
Last time I looked at it, there wasn't any sort of SIP trunk connectivity and the agent connectivity was via a web app. I'd be surprised if they added it. I think you can port a number to them though for the trigger. It's pay per minute the customer call is connected. I imagine they make it easy to hook up with their speech rec/AI stuff a la alexa. On Sat, Oct 6, 2018 at 3:51 PM Lelio Fulgenzi wrote: > > Anyone checking out Amazon Connect? Our call centre software is primarily > for on-prem inbound calls, wondering if they have SIP trunk connectivity so > this can remain to appear as such. Using our existing phones as > agent/remote destinations. > > > *-sent from mobile device-* > > > *Lelio Fulgenzi, B.A.* | Senior Analyst > > Computing and Communications Services | University of Guelph > > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | > N1G 2W1 > > 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca > > > > www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook > > > > [image: University of Guelph Cornerstone with Improve Life tagline] > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Rate-limit voice calls by callerID?
Hi Kent, That's not a bad idea to handle the calling numbers we know about, thanks for the tip. I'm trying to figure out something more generic for when they start picking new numbers so that it doesn't require manual intervention. On Mon, Sep 24, 2018 at 4:40 PM Kent Roberts wrote: > Create a dial peer on cube to match the calling number, then use the call > limits in the dial peer to limit the number. > > Don’t forget to make it a lower preference, then your main ones, or it > will never hit.Hunt stop in the dial peer so it doesn’t roll over…. > > Test after hours with your cell phone or something… > > > > You could also use a TCL script if you have that knowledge. > > > > > > On Sep 24, 2018, at 2:12 PM, Ed Leatherman > wrote: > > > > Hi everyone, > > > > This is a shot in the dark here but has anyone seen a product that can > rate-limit SIP calls based on the calling party number? > > > > We've had a few recent incidents where a scammer has put our telephone > exchange into their robocall system and blasted us with presumably > thousands of call attempts spoofing a government or foreign agency's > Washington DC phone number, for example, the VA or the Chinese Embassy. > Aside from the nuisance of the robocalls, this crushes our SIP concurrent > call paths until I block the number on CUBE. I was thinking if there were a > way to rate limit calls from a calling party number that would probably let > the legitimate calls in, for the most part, but squelch anything that would > DOS us. This doesn't seem to be in the normal CUBE toolset so to speak. > > > > I'm looking at a company called SecureLogix that might be able to do > something like this. Are there any actual SBCs that can handle something > like that? > > > > Any other solutions to this issue? > > > > -- > > Ed Leatherman > > ___ > > cisco-voip mailing list > > cisco-voip@puck.nether.net > > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Rate-limit voice calls by callerID?
Hi everyone, This is a shot in the dark here but has anyone seen a product that can rate-limit SIP calls based on the calling party number? We've had a few recent incidents where a scammer has put our telephone exchange into their robocall system and blasted us with presumably thousands of call attempts spoofing a government or foreign agency's Washington DC phone number, for example, the VA or the Chinese Embassy. Aside from the nuisance of the robocalls, this crushes our SIP concurrent call paths until I block the number on CUBE. I was thinking if there were a way to rate limit calls from a calling party number that would probably let the legitimate calls in, for the most part, but squelch anything that would DOS us. This doesn't seem to be in the normal CUBE toolset so to speak. I'm looking at a company called SecureLogix that might be able to do something like this. Are there any actual SBCs that can handle something like that? Any other solutions to this issue? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CER default call back
What you read was what i understood it to be, so i'm interested in hearing what you find out. On Tue, Jul 31, 2018, 2:26 PM Scott Voll wrote: > on CER 11.5 where does CER send call back numbers after the 3 hours after > the post 911 call? > > I've done some googling and it said the default ERL on site contact, but > that 's not what is happening. Need to change it as an end user is getting > the calls. > > TIA > > scott > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 7832 vs 8832 vs 8831
We've also used the Jabra Bluetooth (Speak model I think) in a few small places and its worked out OK as long as the customer keeps it charged. We've got a handful of 7832's around, they seem to work OK, no complaints. No dongles. I've had some consternation about the 8832 adapter types reported from one of our regional campuses... the field tech didn't read closely which one to order and the partner did not know what to order either so they wasted a good bit of time sending the incorrect parts back and forth. I don't have any of that model to comment on directly. I don't like the 8831's. We still have a bunch of 7937's. Hanging on to them until they quit working. On Mon, Jul 30, 2018 at 8:08 AM Pawlowski, Adam wrote: > We've also used a Jabra Bluetooth speaker/mic combo with an 8851 and found > it works for most smaller huddle spaces without too much trouble. They make > larger variants of that - which, yes, they're several hundred dollars, but > these conference sets can be pushing $1k USD. > > It would be cheaper to outfit a lot of smaller spaces with 8851 on a > table, or, mounted nearby with the Bluetooth devices to augement. > > The 8831 has bene okay other than the part where pressing "conf" to have a > conference call sporadically fails. That bug was terminated and I gather > Cisco/Revo aren't releasing new software for it. We haven't had much > complaint with the external keypad - most of the users here came from the > romulan starship polycom units where the just wrap all the cables around it > and throw it in to a drawer until they need it. This works just fine until > a cable breaks and the replacement parts are several hundred dollars as > well. > > Just my feedback. > > Adam > ___________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Chrome 67 with CUIC/Live Data
Thanks for the clarification! On Tue, Jun 5, 2018 at 1:06 PM, Abhiram Kramadhati (akramadh) < akram...@cisco.com> wrote: > Hi Ed, > > > > This is a permanent fix, but need to make sure that any subsequent ES you > apply should contain the fix. Say, if you apply any of the ES’s that are > currently available **today** then this fix is overwritten and you will > need TAC to run the script again. > > So you will either need to: > >- Get the fix from TAC and make sure any ES you install is released >after this patch (so that it contains this fix) >- Install the cop file directly once it is released > > > > In the worst-case scenario where your fix gets overwritten, you can always > have TAC re-run the script to install the fix manually. But after 20th, > best to install the cop file directly. > > > > Regards, > > Abhiram Kramadhati > > Technical Solutions Manager, CCBU > > CCIE Collaboration # 40065 > > > > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 500 series headsets
Ahhh the Internet.. If it can provide some manageability features and possibly help out with audio switching (like Brian said), we'll at least check them out. On Fri, Apr 13, 2018 at 10:50 AM, Lelio Fulgenzi wrote: > I like the part about the guy who thought 51% was a monster discount, only > to be followed up by someone saying, 51% is not a monster discount. lol > > > > --- > > *Lelio Fulgenzi, B.A.* | Senior Analyst > > Computing and Communications Services | University of Guelph > > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | > N1G 2W1 > > 519-824-4120 Ext. 56354 | le...@uoguelph.ca > > > > www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook > > > > [image: University of Guelph Cornerstone with Improve Life tagline] > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Anthony Holloway > *Sent:* Friday, April 13, 2018 10:34 AM > *To:* Ryan Ratliff (rratliff) > *Cc:* cisco-voip list > *Subject:* Re: [cisco-voip] 500 series headsets > > > > Seems like most of the complaints so far are on features for the price, as > compared to established headset vendors out there. When has Cisco ever > been the cheaper option? Seriously, when? I've always thought of Cisco as > the higher cost option, but maybe that's a perception that is incorrect. > > > > On Fri, Apr 13, 2018 at 9:29 AM Ryan Ratliff (rratliff) < > rratl...@cisco.com> wrote: > > I’m using a 532 with a DX80 and have been pretty happy with it. I’ve > gotten good feedback on audio quality and background noise as well. > > > > For those not on CUG these are the headsets. > > https://www.cisco.com/c/en/us/products/collaboration- > endpoints/headset-500-series/index.html > > > > > > -Ryan > > > > On Apr 13, 2018, at 9:56 AM, Brian Meade wrote: > > > > I played around with them at Enterprise Connect. They seemed pretty nice > especially with the phone menu integration to adjust settings and test > audio. I wasn't a fan of how small the in-call indicator is. I think they > should have made that a full ring. > > > > Having the deep integration into Jabber and the phones will be really > nice, but I've heard they are going to open this up to other headset > vendors as well in the future most likely. > > > > For softphones, I always have customers complaining of different software > taking control of the headset and breaking audio for CIPC/Jabber. If this > can address those issues, it's definitely worth the premium price point and > will help me to start pushing customers more towards softphones. > > > > On Fri, Apr 13, 2018 at 8:39 AM, Lelio Fulgenzi wrote: > > > > For those of you with collab user group memberships... an interesting > thread on the new headsets > > > > > Cisco Series 500 Headsets > > *Collaboration Customer Connection (members only)* - View the full > discussion <https://communities.cisco.com/message/287567#287567> > > > > Sent from my iPhone > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] building an API box - what do I need?
I'm interested in the topic as well. I just have a Linux vm where I can load python or perl mods that I might want to run. To answer your last question, for the few things I've ever done with it I always ask for the credentials at run-time, that way it is running as whatever CUCM user is doing the operation (on the off chance it's not me). We just had an ansible workshop here yesterday, I might start using that for managing VG2XX's (code upgrades and the like). Not strictly API related, but ansible might be another thing you'd run on a box like that if you're inclined. On Wed, Mar 21, 2018 at 4:17 PM, Lelio Fulgenzi wrote: > > I'm going to make a (sad) attempt to try my hand at API usage. You know, > running those sweet commands against CUCM or WebEx, etc to make my life > easier. I'm hoping that by starting with some simple examples, I can build > what I need as time goes on. > > My hope is to build scripts that I can run via CLI, but also create simple > web pages with buttons that execute those scripts, with or without > arguments. The webpages themselves will be protected appropriately. > > But I'm starting from scratch. I do have a resource that I'm hoping will > help out, but I'm wondering, if you were building an api box from scratch, > what sort of things would you want (or need) on it to make things run > smoothly? > > I'm guessing if I want to automate any of these scripts, I'll have no > choice but to store the userID/password either in the script or read from a > file. I've asked this before of a few people, but there is no real way to > store a password securely and decrypt it at time of running the script. > > For interactive scripts, either CLI or web form, do you use the same > userID/password, or are you asking for credentials and use those > credentials? > > I know... a big ask. > > > --- > Lelio Fulgenzi, B.A. | Senior Analyst > Computing and Communications Services | University of Guelph > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | > N1G 2W1 > 519-824-4120 Ext. 56354 | le...@uoguelph.ca<mailto:le...@uoguelph.ca> > > www.uoguelph.ca/ccs<http://www.uoguelph.ca/ccs> | @UofGCCS on Instagram, > Twitter and Facebook > > [University of Guelph Cornerstone with Improve Life tagline] > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] session target dns
I'm not going to explicitly call them out but its in debug snippet from previous post :) It's a regional SP, in their defense they have been willing to work with me on it. On Thu, Mar 15, 2018 at 12:41 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Will the SIP provider remain nameless in this thread? ;) > > On Thu, Mar 15, 2018 at 10:58 AM Ed Leatherman > wrote: > >> I get the impression that im the first customer on these new sbc's. >> >> On Thu, Mar 15, 2018, 11:12 AM Anthony Holloway < >> avholloway+cisco-v...@gmail.com> wrote: >> >>> Wow. So you pointed out a flaw in the provider network. Presumably, >>> they were hosting other customers with the same setup; so how in the world >>> was it working for the others? Or maybe you are the beta tester? >>> >>> On Thu, Mar 15, 2018 at 9:54 AM Ed Leatherman >>> wrote: >>> >>>> Follow-up for posterity.. >>>> >>>> I had a feeling this was the case but got some confirmation from TAC: >>>> "This is working as designed when this is an incoming call, the reason >>>> why it works that way is because on the incoming leg the call comes from 1 >>>> specific IP address, and if CUBE does a DNS query for SRV it might resolve >>>> a different IP address than the one where INVITE is coming from and might >>>> cause inbound calls to fail. Instead, if CUBE does DNS query for A record >>>> it will resolve one specific IP address." >>>> >>>> >>>> The service provider changed their SBC to send an IP address in the >>>> Contact field URI which resolved the issue. >>>> >>>> >>>> Ed >>>> >>>> On Thu, Mar 8, 2018 at 2:16 PM, Ed Leatherman >>>> wrote: >>>> >>>>> >>>>> Follow-up to this SRV/CUBE topic.. >>>>> >>>>> Outbound calls work fine with this setup (after I enabled ip >>>>> domain-lookup ;-) ) >>>>> >>>>> For inbound calls, the service provider is using the hostname for the >>>>> SRV record (peer.isc.lumos.net) in the contact field of the invite. >>>>> Apparently, CUBE only does an A record lookup on that field? >>>>> >>>>> 022206: Mar 8 13:44:04 est: //25051/829EEEDD9B28/SIP/Info/ >>>>> verbose/4608/sipSPIProcessContactInfo: Previous Hop >>>>> peer.isc.lumos.net:5060 >>>>> ... >>>>> 022210: Mar 8 13:44:04 est: //-1//SIP/Info/ >>>>> info/8192/sip_dns_type_a__query: DNS query for peer.isc.lumos.net >>>>> and type:1 >>>>> 022211: Mar 8 13:44:04 est: //-1//SIP/Error/ >>>>> sip_dns_type_a_query: >>>>> TYPE A query failed for peer.isc.lumos.net >>>>> 022212: Mar 8 13:44:04 est: //-1//SIP/Error/_ >>>>> send_dns_fail: >>>>> DNS Query for peer.isc.lumos.net failed >>>>> >>>>> CUBE is basically shutting down the call saying it can't resolve the >>>>> contact field. If I put a local host entry for that name using their >>>>> currently active SBC, inbound calls work. Shouldn't CUBE be doing a SRV >>>>> lookup here, or should the service provider send me an hostname instead of >>>>> an SRV in this field? >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Mar 6, 2018 at 2:25 PM, Anthony Holloway < >>>>> avholloway+cisco-v...@gmail.com> wrote: >>>>> >>>>>> Just so you know, they're not going to know if you use SRV records or >>>>>> not, or host records for that matter. They probably only care about two >>>>>> things: >>>>>> >>>>>> 1) They control which peers you send traffic to via DNS updates >>>>>> >>>>>> 2) They receive the proper/expected host portion in your traffic to >>>>>> them >>>>>> >>>>>> For all intents and purposes, the inclusion of a name in the host >>>>>> portion of a SIP URI is separate from the DNS query. >>>>>> >>>>>> The fact that you point your system at a name (or IP for that matter) >>>>>> and that it then becomes the RHS of the URI is nice, but not required. >>>>>> >>>>>> Therefore, i
Re: [cisco-voip] session target dns
I get the impression that im the first customer on these new sbc's. On Thu, Mar 15, 2018, 11:12 AM Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Wow. So you pointed out a flaw in the provider network. Presumably, they > were hosting other customers with the same setup; so how in the world was > it working for the others? Or maybe you are the beta tester? > > On Thu, Mar 15, 2018 at 9:54 AM Ed Leatherman > wrote: > >> Follow-up for posterity.. >> >> I had a feeling this was the case but got some confirmation from TAC: >> "This is working as designed when this is an incoming call, the reason >> why it works that way is because on the incoming leg the call comes from 1 >> specific IP address, and if CUBE does a DNS query for SRV it might resolve >> a different IP address than the one where INVITE is coming from and might >> cause inbound calls to fail. Instead, if CUBE does DNS query for A record >> it will resolve one specific IP address." >> >> >> The service provider changed their SBC to send an IP address in the >> Contact field URI which resolved the issue. >> >> >> Ed >> >> On Thu, Mar 8, 2018 at 2:16 PM, Ed Leatherman >> wrote: >> >>> >>> Follow-up to this SRV/CUBE topic.. >>> >>> Outbound calls work fine with this setup (after I enabled ip >>> domain-lookup ;-) ) >>> >>> For inbound calls, the service provider is using the hostname for the >>> SRV record (peer.isc.lumos.net) in the contact field of the invite. >>> Apparently, CUBE only does an A record lookup on that field? >>> >>> 022206: Mar 8 13:44:04 est: >>> //25051/829EEEDD9B28/SIP/Info/verbose/4608/sipSPIProcessContactInfo: >>> Previous Hop peer.isc.lumos.net:5060 >>> ... >>> 022210: Mar 8 13:44:04 est: >>> //-1//SIP/Info/info/8192/sip_dns_type_a__query: DNS query >>> for peer.isc.lumos.net and type:1 >>> 022211: Mar 8 13:44:04 est: >>> //-1//SIP/Error/sip_dns_type_a_query: >>> TYPE A query failed for peer.isc.lumos.net >>> 022212: Mar 8 13:44:04 est: //-1//SIP/Error/_send_dns_fail: >>> DNS Query for peer.isc.lumos.net failed >>> >>> CUBE is basically shutting down the call saying it can't resolve the >>> contact field. If I put a local host entry for that name using their >>> currently active SBC, inbound calls work. Shouldn't CUBE be doing a SRV >>> lookup here, or should the service provider send me an hostname instead of >>> an SRV in this field? >>> >>> >>> >>> >>> On Tue, Mar 6, 2018 at 2:25 PM, Anthony Holloway < >>> avholloway+cisco-v...@gmail.com> wrote: >>> >>>> Just so you know, they're not going to know if you use SRV records or >>>> not, or host records for that matter. They probably only care about two >>>> things: >>>> >>>> 1) They control which peers you send traffic to via DNS updates >>>> >>>> 2) They receive the proper/expected host portion in your traffic to them >>>> >>>> For all intents and purposes, the inclusion of a name in the host >>>> portion of a SIP URI is separate from the DNS query. >>>> >>>> The fact that you point your system at a name (or IP for that matter) >>>> and that it then becomes the RHS of the URI is nice, but not required. >>>> >>>> Therefore, if you ask them to commit to telling you about IP address >>>> changes completely negates their desire to use SRV records. Just say'n. >>>> >>>> On Tue, Mar 6, 2018 at 6:30 AM Ed Leatherman >>>> wrote: >>>> >>>>> Thanks Anthony, That was spot on what I was trying to figure out. I've >>>>> been using server-groups up until now (and will continue on the CUCM >>>>> facing >>>>> side), the service provider is forcing the change on the side facing them. >>>>> >>>>> Loren: That's an interesting idea to lock in the host resolution on >>>>> the CUBE itself, but in this case I think it might set me up for an outage >>>>> if the service provider changes their IP Addressing. Maybe I can get them >>>>> to commit to telling me before they change those.. >>>>> >>>>> On Mon, Mar 5, 2018 at 2:31 PM, Anthony Holloway < >>>>> avholloway+cisco-v...@gmail.com> wrote: >>>>
Re: [cisco-voip] session target dns
Follow-up for posterity.. I had a feeling this was the case but got some confirmation from TAC: "This is working as designed when this is an incoming call, the reason why it works that way is because on the incoming leg the call comes from 1 specific IP address, and if CUBE does a DNS query for SRV it might resolve a different IP address than the one where INVITE is coming from and might cause inbound calls to fail. Instead, if CUBE does DNS query for A record it will resolve one specific IP address." The service provider changed their SBC to send an IP address in the Contact field URI which resolved the issue. Ed On Thu, Mar 8, 2018 at 2:16 PM, Ed Leatherman wrote: > > Follow-up to this SRV/CUBE topic.. > > Outbound calls work fine with this setup (after I enabled ip domain-lookup > ;-) ) > > For inbound calls, the service provider is using the hostname for the SRV > record (peer.isc.lumos.net) in the contact field of the invite. > Apparently, CUBE only does an A record lookup on that field? > > 022206: Mar 8 13:44:04 est: > //25051/829EEEDD9B28/SIP/Info/verbose/4608/sipSPIProcessContactInfo: > Previous Hop peer.isc.lumos.net:5060 > ... > 022210: Mar 8 13:44:04 est: //-1//SIP/Info/ > info/8192/sip_dns_type_a__query: DNS query for peer.isc.lumos.net and > type:1 > 022211: Mar 8 13:44:04 est: //-1//SIP/Error/ > sip_dns_type_a_query: > TYPE A query failed for peer.isc.lumos.net > 022212: Mar 8 13:44:04 est: //-1//SIP/Error/_send_dns_fail: > DNS Query for peer.isc.lumos.net failed > > CUBE is basically shutting down the call saying it can't resolve the > contact field. If I put a local host entry for that name using their > currently active SBC, inbound calls work. Shouldn't CUBE be doing a SRV > lookup here, or should the service provider send me an hostname instead of > an SRV in this field? > > > > > On Tue, Mar 6, 2018 at 2:25 PM, Anthony Holloway < > avholloway+cisco-v...@gmail.com> wrote: > >> Just so you know, they're not going to know if you use SRV records or >> not, or host records for that matter. They probably only care about two >> things: >> >> 1) They control which peers you send traffic to via DNS updates >> >> 2) They receive the proper/expected host portion in your traffic to them >> >> For all intents and purposes, the inclusion of a name in the host portion >> of a SIP URI is separate from the DNS query. >> >> The fact that you point your system at a name (or IP for that matter) and >> that it then becomes the RHS of the URI is nice, but not required. >> >> Therefore, if you ask them to commit to telling you about IP address >> changes completely negates their desire to use SRV records. Just say'n. >> >> On Tue, Mar 6, 2018 at 6:30 AM Ed Leatherman >> wrote: >> >>> Thanks Anthony, That was spot on what I was trying to figure out. I've >>> been using server-groups up until now (and will continue on the CUCM facing >>> side), the service provider is forcing the change on the side facing them. >>> >>> Loren: That's an interesting idea to lock in the host resolution on the >>> CUBE itself, but in this case I think it might set me up for an outage if >>> the service provider changes their IP Addressing. Maybe I can get them to >>> commit to telling me before they change those.. >>> >>> On Mon, Mar 5, 2018 at 2:31 PM, Anthony Holloway < >>> avholloway+cisco-v...@gmail.com> wrote: >>> >>>> Loren, >>>> >>>> Just out of curiosity, why didn't you just use session server groups? >>>> Based on the config you shared, it looks like it would achieve the same >>>> thing, but with less config, and not adding in the DNS stack within IOS. >>>> >>>> Ed, >>>> >>>> *Note, you cannot use DNS in server groups, so it's one or the other. >>>> >>>> I also think it's important to know that the IOS code is written such >>>> that it will look for SRV records first, and then fallback to looking for >>>> an A (host) record once the DNS timeouts. >>>> >>>> E.g., >>>> >>>> You enter "session target dns:collab.domain.com" >>>> >>>> IOS looks for _sip._udp.collab.domain.com SRV record first, timesout, >>>> then looks for collab.domain.com host record second. >>>> >>>> *Note that the outgoing session transport on IOS is UDP by default, >>>> unless you change it to TCP with the command &
Re: [cisco-voip] session target dns
Follow-up to this SRV/CUBE topic.. Outbound calls work fine with this setup (after I enabled ip domain-lookup ;-) ) For inbound calls, the service provider is using the hostname for the SRV record (peer.isc.lumos.net) in the contact field of the invite. Apparently, CUBE only does an A record lookup on that field? 022206: Mar 8 13:44:04 est: //25051/829EEEDD9B28/SIP/Info/verbose/4608/sipSPIProcessContactInfo: Previous Hop peer.isc.lumos.net:5060 ... 022210: Mar 8 13:44:04 est: //-1//SIP/Info/info/8192/sip_dns_type_a__query: DNS query for peer.isc.lumos.net and type:1 022211: Mar 8 13:44:04 est: //-1//SIP/Error/sip_dns_type_a_query: TYPE A query failed for peer.isc.lumos.net 022212: Mar 8 13:44:04 est: //-1//SIP/Error/_send_dns_fail: DNS Query for peer.isc.lumos.net failed CUBE is basically shutting down the call saying it can't resolve the contact field. If I put a local host entry for that name using their currently active SBC, inbound calls work. Shouldn't CUBE be doing a SRV lookup here, or should the service provider send me an hostname instead of an SRV in this field? On Tue, Mar 6, 2018 at 2:25 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Just so you know, they're not going to know if you use SRV records or not, > or host records for that matter. They probably only care about two things: > > 1) They control which peers you send traffic to via DNS updates > > 2) They receive the proper/expected host portion in your traffic to them > > For all intents and purposes, the inclusion of a name in the host portion > of a SIP URI is separate from the DNS query. > > The fact that you point your system at a name (or IP for that matter) and > that it then becomes the RHS of the URI is nice, but not required. > > Therefore, if you ask them to commit to telling you about IP address > changes completely negates their desire to use SRV records. Just say'n. > > On Tue, Mar 6, 2018 at 6:30 AM Ed Leatherman > wrote: > >> Thanks Anthony, That was spot on what I was trying to figure out. I've >> been using server-groups up until now (and will continue on the CUCM facing >> side), the service provider is forcing the change on the side facing them. >> >> Loren: That's an interesting idea to lock in the host resolution on the >> CUBE itself, but in this case I think it might set me up for an outage if >> the service provider changes their IP Addressing. Maybe I can get them to >> commit to telling me before they change those.. >> >> On Mon, Mar 5, 2018 at 2:31 PM, Anthony Holloway < >> avholloway+cisco-v...@gmail.com> wrote: >> >>> Loren, >>> >>> Just out of curiosity, why didn't you just use session server groups? >>> Based on the config you shared, it looks like it would achieve the same >>> thing, but with less config, and not adding in the DNS stack within IOS. >>> >>> Ed, >>> >>> *Note, you cannot use DNS in server groups, so it's one or the other. >>> >>> I also think it's important to know that the IOS code is written such >>> that it will look for SRV records first, and then fallback to looking for >>> an A (host) record once the DNS timeouts. >>> >>> E.g., >>> >>> You enter "session target dns:collab.domain.com" >>> >>> IOS looks for _sip._udp.collab.domain.com SRV record first, timesout, >>> then looks for collab.domain.com host record second. >>> >>> *Note that the outgoing session transport on IOS is UDP by default, >>> unless you change it to TCP with the command "session transport tcp" at the >>> "voice service voip" level, or at the dial-peer level. So, having a >>> _sip._tcp SRV record on your CUBE would create a confusing scenario. >>> Contrast this with the incoming connection, which can be either. However, >>> SRV records, like Loren is showing, are for outbound connection >>> establishments. >>> >>> I have not done an extensive amount of testing here, but I would be >>> curious to know if IOS handles the TTL for the DNS record correctly, or if >>> it queries DNS for every setup like how that one defect was hitting CUCM >>> SIP trunks for a while there. I looked for "TTL" in the CVP Config guide, >>> but it didn't say. >>> >>> On Mon, Mar 5, 2018 at 11:19 AM Loren Hillukka >>> wrote: >>> >>>> You can have your gw query your DNS server, and you have to add SRV >>>> records to your central DNS server (like with the jabber entries re
Re: [cisco-voip] session target dns
Thanks Ryan (but who really knows when it comes to IOS bugs). +1 Here is the bug condition: > > > > *Upon start-up/reboot the DNS process doesn't initiate a query till around > 18 minutes after the boot up. This long delay results in hostname > configured features (ex:NTP servers) not being used till this process is > complete. Even when doing this time the DNS server is reachable.* > > > > Thanks, > > > > Ryan > > *From: *Ed Leatherman > *Sent: *Tuesday, March 6, 2018 7:31 AM > *To: *Anthony Holloway > *Cc: *Cisco VOIP > *Subject: *Re: [cisco-voip] session target dns > > > > Thanks Anthony, That was spot on what I was trying to figure out. I've > been using server-groups up until now (and will continue on the CUCM facing > side), the service provider is forcing the change on the side facing them. > > > > Loren: That's an interesting idea to lock in the host resolution on the > CUBE itself, but in this case I think it might set me up for an outage if > the service provider changes their IP Addressing. Maybe I can get them to > commit to telling me before they change those.. > > > > On Mon, Mar 5, 2018 at 2:31 PM, Anthony Holloway < > avholloway+cisco-v...@gmail.com> wrote: > > Loren, > > > > Just out of curiosity, why didn't you just use session server groups? > Based on the config you shared, it looks like it would achieve the same > thing, but with less config, and not adding in the DNS stack within IOS. > > > > Ed, > > > > *Note, you cannot use DNS in server groups, so it's one or the other. > > > > I also think it's important to know that the IOS code is written such that > it will look for SRV records first, and then fallback to looking for an A > (host) record once the DNS timeouts. > > > > E.g., > > > > You enter "session target dns:collab.domain.com" > > > > IOS looks for _sip._udp.collab.domain.com SRV record first, timesout, > then looks for collab.domain.com host record second. > > > > *Note that the outgoing session transport on IOS is UDP by default, unless > you change it to TCP with the command "session transport tcp" at the "voice > service voip" level, or at the dial-peer level. So, having a _sip._tcp SRV > record on your CUBE would create a confusing scenario. Contrast this with > the incoming connection, which can be either. However, SRV records, like > Loren is showing, are for outbound connection establishments. > > > > I have not done an extensive amount of testing here, but I would be > curious to know if IOS handles the TTL for the DNS record correctly, or if > it queries DNS for every setup like how that one defect was hitting CUCM > SIP trunks for a while there. I looked for "TTL" in the CVP Config guide, > but it didn't say. > > > > On Mon, Mar 5, 2018 at 11:19 AM Loren Hillukka > wrote: > > You can have your gw query your DNS server, and you have to add SRV > records to your central DNS server (like with the jabber entries required > to get jabber sign-in to work). > > Here’s the example of doing local DNS to static entries on the gateway > itself, from the CVP 10 config guide. CVP is where I first started doing > dns srv on the local gateway, as I preferred breaking the call center > myself instead of having the AD/DNS teams do it for me without me knowing > ;-) > > === > > You can also configure the Gateway statically instead of using DNS. The > following example shows how both the A and SRV type records could be > configured: > > ip host cvp4cc2.cisco.com 10.4.33.132 > > ip host cvp4cc3.cisco.com 10.4.33.133 > > ip host cvp4cc1.cisco.com 10.4.33.131 > > For SIP/TCP: > > ip host _sip._tcp.cvp.cisco.com srv 50 50 5060cvp4cc3.cisco.com > > ip host _sip._tcp.cvp.cisco.com srv 50 50 5060cvp4cc2.cisco.com > > ip host _sip._tcp.cvp.cisco.com srv 50 50 5060cvp4cc1.cisco.com > > For SIP/UDP: > > ip host _sip._udp.cvp.cisco.com srv 50 50 5060cvp4cc3.cisco.com > > ip host _sip._udp.cvp.cisco.com srv 50 50 5060cvp4cc2.cisco.com > > ip host _sip._udp.cvp.cisco.com srv 50 50 5060cvp4cc1.cisco.com > > > > Then your dial-peer would have session target dns:cvp.cisco.com which > would point to the SRV record, which would use the weight/priority values > to choose the final host, and resolve the selected host to an IP using the > normal "ip host name x.x.x.x" entry > > > > > > Loren > > > On Mar 5, 2018, at 10:15 AM, Ed Leatherman wrote: > > Hi everyone, > > > > Hopefully a quick question - in a dial-peer on
Re: [cisco-voip] cisco-voip Digest, Vol 173, Issue 3
Thanks for the tip! Q On Mon, Mar 5, 2018, 3:58 PM Loren Hillukka wrote: > Nice tips Adam. The failovers to active endpoints was a pain. That retry > invites 2 was a must - the default was 6. > > Loren > > > On Mar 5, 2018, at 1:14 PM, Pawlowski, Adam wrote: > > > > Ed, > > > > Caveat on all this that I set this up a year or two ago so I could be > wrong on some parts: > > > > This does DNS SRV lookup I believe first, then A. As long as you're in > an IOS version that supports SRV as that appeared somewhere in 15 I > believe. We are running 15.(4)3 M2. > > > > I set up a number of SRV records to point at my UCM cluster, for > example, and then have the dial-peer set to: > > > > session target dns:prod-all.cmgroup.subzone.zone.internal > > > > I don't see that I have specified "session transport tcp" in this router > but this is a _SIP._TCP SRV record and it is up and running. > > > > I will say that you want to be careful with using this with the > busyout/options ping. In my experience when it pulls the various SRV hosts, > with respect to preference and weight, it will pick the one it should be > using at any given point in time but it will not test all UCM nodes. It > could be possible to busyout the group if multiple failures were to occur > for some reason, even if other hosts are up, based on how IOS handles the > responses to this. I sort of recall watching it, and test #1 may go to subA > which fails, but then test #2 goes to subB so you're good. If you'd set the > weighting different you could knock the peer offline. > > > > I think. > > > > I know that in response to this behavior I did add this configuration to > my sip user agent configuration: > > > > sip-ua > > retry invite 2 > > timers trying 100 > > ! > > > > This was to cause the router to try again (perhaps to a different peer) > if it failed to get anywhere with one. The defaults were too long for the > call to fail out to a particular peer before exceeding network timers and > losing the call. > > > > Best, > > > > Adam Pawlowski > > SUNYAB NCS > > > > ___ > > cisco-voip mailing list > > cisco-voip@puck.nether.net > > https://puck.nether.net/mailman/listinfo/cisco-voip > ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] session target dns
Thanks Anthony, That was spot on what I was trying to figure out. I've been using server-groups up until now (and will continue on the CUCM facing side), the service provider is forcing the change on the side facing them. Loren: That's an interesting idea to lock in the host resolution on the CUBE itself, but in this case I think it might set me up for an outage if the service provider changes their IP Addressing. Maybe I can get them to commit to telling me before they change those.. On Mon, Mar 5, 2018 at 2:31 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Loren, > > Just out of curiosity, why didn't you just use session server groups? > Based on the config you shared, it looks like it would achieve the same > thing, but with less config, and not adding in the DNS stack within IOS. > > Ed, > > *Note, you cannot use DNS in server groups, so it's one or the other. > > I also think it's important to know that the IOS code is written such that > it will look for SRV records first, and then fallback to looking for an A > (host) record once the DNS timeouts. > > E.g., > > You enter "session target dns:collab.domain.com" > > IOS looks for _sip._udp.collab.domain.com SRV record first, timesout, > then looks for collab.domain.com host record second. > > *Note that the outgoing session transport on IOS is UDP by default, unless > you change it to TCP with the command "session transport tcp" at the "voice > service voip" level, or at the dial-peer level. So, having a _sip._tcp SRV > record on your CUBE would create a confusing scenario. Contrast this with > the incoming connection, which can be either. However, SRV records, like > Loren is showing, are for outbound connection establishments. > > I have not done an extensive amount of testing here, but I would be > curious to know if IOS handles the TTL for the DNS record correctly, or if > it queries DNS for every setup like how that one defect was hitting CUCM > SIP trunks for a while there. I looked for "TTL" in the CVP Config guide, > but it didn't say. > > On Mon, Mar 5, 2018 at 11:19 AM Loren Hillukka > wrote: > >> You can have your gw query your DNS server, and you have to add SRV >> records to your central DNS server (like with the jabber entries required >> to get jabber sign-in to work). >> >> Here’s the example of doing local DNS to static entries on the gateway >> itself, from the CVP 10 config guide. CVP is where I first started doing >> dns srv on the local gateway, as I preferred breaking the call center >> myself instead of having the AD/DNS teams do it for me without me knowing >> ;-) >> >> === >> >> You can also configure the Gateway statically instead of using DNS. The >> following example shows how both the A and SRV type records could be >> configured: >> >> ip host cvp4cc2.cisco.com 10.4.33.132 >> >> ip host cvp4cc3.cisco.com 10.4.33.133 >> >> ip host cvp4cc1.cisco.com 10.4.33.131 >> >> For SIP/TCP: >> >> ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc3.cisco.com >> >> ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc2.cisco.com >> >> ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc1.cisco.com >> >> For SIP/UDP: >> >> ip host _sip._udp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc3.cisco.com >> >> ip host _sip._udp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc2.cisco.com >> >> ip host _sip._udp.cvp.cisco.com srv 50 50 5060 <50%2050%205060> >> cvp4cc1.cisco.com >> >> >> >> Then your dial-peer would have session target dns:cvp.cisco.com which >> would point to the SRV record, which would use the weight/priority values >> to choose the final host, and resolve the selected host to an IP using the >> normal "ip host name x.x.x.x" entry >> >> >> Loren >> >> On Mar 5, 2018, at 10:15 AM, Ed Leatherman >> wrote: >> >> Hi everyone, >> >> Hopefully a quick question - in a dial-peer on CUBE (16.3.5) how does >> session target dns: resolve to an IP? I've never used DNS as target before >> for this. >> >> Does CUBE just do a query for the A record by default, or does it do a >> SRV query by default? I have a SIP provider that wants to start using SRV >> for their SBC(s) and I'm researching how to setup my end in IOS. If it >> doesn't query SRV default, where do I toggle that
[cisco-voip] session target dns
Hi everyone, Hopefully a quick question - in a dial-peer on CUBE (16.3.5) how does session target dns: resolve to an IP? I've never used DNS as target before for this. Does CUBE just do a query for the A record by default, or does it do a SRV query by default? I have a SIP provider that wants to start using SRV for their SBC(s) and I'm researching how to setup my end in IOS. If it doesn't query SRV default, where do I toggle that behavior? The command reference just says "Configures the host device housing the domain name system (DNS) server that resolves the name of the dial peer to receive calls." I've found the knob to tell it what SRV format to use in the sip-ua section. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] anyone try solaborate hello?
you're on the customer side, is it different for you? Do you > have the freedom of just buying and trying any product you want, regardless > of manufacturer? Even when I was on the customer side for a few years, I > was still a Cisco Engineer, and only worked on Cisco products. Perhaps you > wear more hats than just Cisco Engineer, and that's our difference. > > > > Anyway, just curious. > > > > By the way, this product actually looks pretty cool and he price is very > affordable. So, my above comments are not tied to the product itself, more > to the idea of 3rd party COTS products competing against Cisco products, > when viewed from a Cisco Engineer's perspective. > > On Wed, Jan 31, 2018 at 2:13 PM Lelio Fulgenzi wrote: > > Has anyone tried this product? > > https://www.solaborate.com/hello > > > > --- > Lelio Fulgenzi, B.A. | Senior Analyst > Computing and Communications Services | University of Guelph > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | > N1G 2W1 > 519-824-4120 Ext. 56354 <(519)%20824-4120> | le...@uoguelph.ca le...@uoguelph.ca> > > www.uoguelph.ca/ccs<http://www.uoguelph.ca/ccs> | @UofGCCS on Instagram, > Twitter and Facebook > > [University of Guelph Cornerstone with Improve Life tagline] > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Spectre and Meltdown remediation as relevant to Cisco systems
>From what info I'm aware of, hypervisor fixes (at least vmware) are not resulting in a perceptible performance degradation, however fixes at the guest OS level are showing performance issues depending on the type of operation involved. To completely mitigate the vulnerabilities, seems like in most cases it requires a multi-faceted effort, BIOS/firmware/CPU, Hypervisor (if present), and OS all must be updated to address all of the attack vectors. Right now the fixes at the OS layer don't see fully baked. I feel like if you're 100% appliance based VM's wrt Cisco UC apps and they are the only things running in the cluster, your risk is pretty low and letting details/patches get sussed out is logical before you go crazy patching things. If there are non-UC or non-appliance items running in the same cluster, then addressing at the hardware and hypervisor level is important, followed by guest OS fixes for those other VMs once you understand the impact on those. Just my current thinking anyway. I bet we don't see any UCOS patches that address this at the OS level until its fully baked or its just part of the linux kernel they use. On Tue, Jan 9, 2018 at 8:32 PM, Lelio Fulgenzi wrote: > > To be honest, I'm a little worried about the rumoured slowdown the fixes > are gonna have. Will this impact the supported status of certain CPUs in > collab suite? > > Sent from my iPhone > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] records management software for phones, switches, fiber/cable plant
Service is @ https://www.calero.com/pinnacle/ , its actually a calero product now. On Thu, Oct 26, 2017 at 1:06 PM, Ed Leatherman wrote: > We're using a SaaS product called Pinnacle that can do those things (i'm > 99% sure on the cable plant stuff), although we're only using it for > charge-back internally. > > On Thu, Oct 26, 2017 at 12:15 PM, James Conyers > wrote: > >> Hi, >> >> Does anyone have an “all inclusive” records management solution that they >> would recommend to help keep records for things like: >> >> · Cisco phones – who the phone belongs to, what room number, mac >> address, etc >> >> · Cisco switches – where the switches are located, what model, >> port descriptions, etc >> >> · Cable plant inventory – both fiber optic cables as well as >> copper, type of cable, terminations, strands, bulkheads, etc >> >> >> >> Any help on one or more products that can be used either separately, or >> in conjunction with each other is greatly appreciated! >> >> >> >> Thanks, >> >> James >> >> >> >> James Conyers >> >> Telecommunications Engineer II >> >> 303-871-7992 <(303)%20871-7992> >> >> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > > -- > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] records management software for phones, switches, fiber/cable plant
We're using a SaaS product called Pinnacle that can do those things (i'm 99% sure on the cable plant stuff), although we're only using it for charge-back internally. On Thu, Oct 26, 2017 at 12:15 PM, James Conyers wrote: > Hi, > > Does anyone have an “all inclusive” records management solution that they > would recommend to help keep records for things like: > > · Cisco phones – who the phone belongs to, what room number, mac > address, etc > > · Cisco switches – where the switches are located, what model, > port descriptions, etc > > · Cable plant inventory – both fiber optic cables as well as > copper, type of cable, terminations, strands, bulkheads, etc > > > > Any help on one or more products that can be used either separately, or in > conjunction with each other is greatly appreciated! > > > > Thanks, > > James > > > > James Conyers > > Telecommunications Engineer II > > 303-871-7992 <(303)%20871-7992> > > > > _______ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] python project ideas
IIRC Unity Connection has a REST API as well On Fri, Oct 13, 2017 at 12:11 PM, Abebe Amare wrote: > Thanks for your input guys. > > I will spin up CUCM and expressway VMs and get working. > > best regards, > > Abebe > > On Fri, Oct 13, 2017 at 12:31 PM, Heim, Dennis > wrote: > >> Expressway API’s should be fairly easy. >> >> >> >> *Dennis Heim | Emerging Technology Architect (Collaboration)* >> >> World Wide Technology, Inc. | +1 314-212-1814 <+1%20314-212-1814> >> >> [image: cid:image001.png@01D10DD2.7FC81F90] >> <https://twitter.com/CollabSensei> >> >> [image: cid:image002.png@01D10DD2.7FC81F90][image: >> cid:image003.png@01D10DD2.7FC81F90] <+13142121814>[image: >> cid:image004.png@01D10DD2.7FC81F90] >> >> "Worry less about who you might offend, and more about who you might >> inspire" -- Tim Allen >> >> “When you have unlimited time, its easy” – Captain Chesley Sullenberger >> >> “There is a fine line between Wrong and Visionary. Unfortunately, you >> have to be a visionary to see it." – Sheldon Cooper >> >> “The greatest danger for most of us is not that our aim is too high and >> we miss it, but that it is too low and we reach it.” -- Michelangelo >> Buonarroti >> >> “We should transform the way we work” – Rowan Trollope >> >> “If you’re not failing every now and again, it’s a sign you’re not doing >> anything very innovative” – Woody Allen >> >> >> >> *Click here to join me in my Collaboration Meeting Room >> <https://wwt.webex.com/meet/dennis.heim>* >> >> >> >> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On >> Behalf Of *Abebe Amare >> *Sent:* Friday, October 13, 2017 11:13 AM >> *To:* cisco voip >> *Subject:* [cisco-voip] python project ideas >> >> >> >> Hi, >> >> >> >> I am taking Python for network engineers course and have to do a project >> on it. The project must be using an API or a major framework to do >> something. It ideally should involve manipulating real-world data. >> >> I was looking to do my project focusing on Cisco collaboration using REST >> API. Can you guys suggest any project ideas? >> >> >> >> Thanks in advance >> > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PLM dead
At CiscoLive this year it sounded like they were getting push from top down to unify product licensing, so hopefully CC gets on the same boat. Not thrilled that its all online but if it gets everything consistent that will be a plus. On Fri, Oct 13, 2017 at 10:44 AM, Lelio Fulgenzi wrote: > > > I knew that v12 was smart licensing only, but I did not know that they > were removing PLM. I’m guessing it’s going to follow the rest of Cisco > products (finally) and use whatever smart licensing deployment you have on > site. They have a few options, we’ll likely go with a proxy. > > > > Of course, UCCx will use it’s own licensing solution. Maybe even pick up > PLM at a firesale. ;) > > > > --- > > Lelio Fulgenzi, B.A. > > Senior Analyst, Network Infrastructure > > Computing and Communications Services (CCS) > > University of Guelph > > > > 519-824-4120 Ext 56354 <(519)%20824-4120> > > le...@uoguelph.ca > > www.uoguelph.ca/ccs > > Room 037, Animal Science and Nutrition Building > > Guelph, Ontario, N1G 2W1 > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Scott Voll > *Sent:* Friday, October 13, 2017 10:31 AM > *To:* cisco-voip@puck.nether.net > *Subject:* [cisco-voip] PLM dead > > > > am I last to know? They are once again moving licensing? > > > > https://communities.cisco.com/message/271653 > > > > Scott > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Expert-Level Cert Renewal
Looks like there's a $300 admin fee (basically what you'd pay for the written exam i think), and then options for credits, including cisco live sessions which is kinda nice. On Tue, Jun 6, 2017 at 11:56 AM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > This seems like great news, as I loath the current CCIE Collaboration > Written exam. Which, I have by the way, submitted an application to help > improve it, but no one has ever gotten back to me on. > > I'll look into this more, but as long as there's no catch (like $5,000 in > cost), this is a great alternative to taking the written. > > On Tue, Jun 6, 2017 at 8:13 AM Daniel Pagan wrote: > >> Some of you might be interested to hear that Cisco has announced a new >> method of renewing one’s expert-level certification. While taking a written >> exam is still an option, those needing to renew their CCIE/CCDE will be >> allowed to enroll themselves in select classes, online and in-person, which >> will count towards renewal credits. Renewal will require 100 credits in >> total and applies to both active and suspended status cert holders. >> >> >> >> More information: https://learningnetwork.cisco. >> com/community/certifications/cisco-continuing-education-program >> >> >> >> HTH >> >> >> >> - Dan >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 10.6.1 SU2 and VMware
Jose, are you upgrading just vCenter, or are you upgrading vCenter as well as your vSphere version on the esxi hosts? You could likely run vCenter 6.5 and run vSphere 5.5 or 6.0 on your hosts that have UCCX on them. On Wed, Apr 5, 2017 at 11:12 PM, Abhiram Kramadhati (akramadh) < akram...@cisco.com> wrote: > UCCX 11.6 will support ESXi 6.5 > > > > Regards, > > Abhiram Kramadhati > > Technical Solutions Manager, CCBU > > CCIE Collaboration # 40065 > > > > > > > > *From: *cisco-voip on behalf of Jose > Colon II > *Reply-To: *"jcolon...@gmail.com" > *Date: *Thursday, 6 April 2017 at 2:19 AM > *To: *Matthew Loraditch > *Cc: *Cisco VOIP > *Subject: *Re: [cisco-voip] UCCX 10.6.1 SU2 and VMware > > > > Thank you for that! Appreciate it. > > > > On Wed, Apr 5, 2017 at 3:46 PM, Matthew Loraditch heliontechnologies.com> wrote: > > ESXi 6.5 is not yet compatible with any Cisco Voice software. > > http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/uc_ > system/virtualization/cisco-collaboration-virtualization.html > > > > Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA > Network Engineer > Direct Voice: 443.541.1518 <(443)%20541-1518> > > Facebook <https://www.facebook.com/heliontech?ref=hl> | Twitter > <https://twitter.com/HelionTech> | LinkedIn > <https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> | > G+ <https://plus.google.com/+Heliontechnologies/posts> > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Jose Colon II > *Sent:* Wednesday, April 5, 2017 4:41 PM > *To:* Cisco VOIP > *Subject:* [cisco-voip] UCCX 10.6.1 SU2 and VMware > > > > We are going to be upgrading our VMware environment from vCenter 5.5 to > 6.5 and can not find any specific information on compatibility with UCCX > 10.6.1 SU2 for this. Is this compatible? > > > > Any help with this is appreciated. > > > > Thanks > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Local Number Portability / Timeframes
Hello! I'm working on porting some telephone numbers and I've been running into a roadblock with one of the carriers (no surprise). Any telco policy guru's here? We've got about 100 numbers we want to port this time, and keep getting kick back from the winning carrier that they aren't able to get the CSR information from the losing carrier, so right now we're trying to play go-between; this isnt going to fly long-term though. I have about 6000 or so I need to port once we get the process down. It seems there are plenty of FCC ruling around "Simple" number ports, but I'm having some trouble finding what the definition of "Simple" is (still looking) and also what if any the requirements are for ports that are more like "Bulk". I get the impression that "Simple" is single numbers, for example a residential customer moving their home service. Anyone have experience with porting over larger chunks of numbers between carriers? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] n-1 metric
Varies per app and per version, some combinations just put you in a bind. We're running UCCX 10.6 with CUCM 11 currently. On Thu, Feb 2, 2017 at 4:58 PM, Scott Voll wrote: > is there an n-1 type metric so you know what you can upgrade before CM? > > we have CER, UCCX, UC, PLM, CUPS, and CM. > > Looks like UCCX and CM have to upgrade together (thought that was > changing, but guess not), but looking for a doc on what can be done before > hand. Just trying to have a few smaller maintenance windows rather than > one big one. > > currently 10.5 moving to 11.5 > > TIA > > Scott > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco 8851 and Bluetooth Speaker
We've been using the speak 510's for just this reason and they've worked out OK.. actually started using them with 8945's initially but bluetooth seems to work better on the 8800's, anecdotally. I haven't been thrilled with the 8831. Internal customers don't care for the form-factor and they haven't been as reliable as the 7937's for us. On Fri, Jan 20, 2017 at 10:02 AM, Pawlowski, Adam wrote: > Has anyone tried to use the Cisco 8851 and a Bluetooth speakerphone as > sort of a poor man's conferencing set? The sets themselves sound pretty > good on their own, but, something like the Jabra Speak 510 that is > Bluetooth enabled would be a more cost effective option than the $900+ 8831 > set (which still has an unresolved bug causing the Conf button to randomly > fail anyways). > > It is obviously not explicitly called out as supported, but, anyone tried > this and had any success? Or, any recommendation on a 3rd party > conferencing phone that is more cost effective, even if not as capable, for > a smaller (6 - 8 person) conference? > > Regards, > > Adam Pawlowski > SUNYAB NCS > aj...@buffalo.edu > +1.716.6458489 > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Does CCX enforce licensing?
If you are savvy with doing custom reports, it looks like there might be a stored procedure on UCCX 9 that you might be able to access - reference: https://supportforums.cisco.com/discussion/12309471/uccx-licenseunit-consumption Otherwise i think you would need to monitor the real time report during your busy period and estimate. On Tue, Nov 29, 2016 at 9:45 AM, Ben Amick wrote: > What about on v9 with only the core CCX and Historical Reports? We don’t > have CUIC, though we’re looking to get it when we upgrade > > > > *Ben Amick* > > Telecom Analyst > > Phone: 216-426-3535 x1104 > > 1457 East 40th Street | Cleveland, OH 44103 > > bam...@humanarc.com | www.humanarc.com > <http://www.humanarc.com%20> > > [image: cid:image001.gif@01CEF0F8.9227CA80] > > > > *From:* Ed Leatherman [mailto:ealeather...@gmail.com] > *Sent:* Tuesday, November 29, 2016 9:41 AM > *To:* Ben Amick > *Cc:* James Buchanan ; > cisco-voip@puck.nether.net > > *Subject:* Re: [cisco-voip] Does CCX enforce licensing? > > > > There is a license utilization hourly report in CUIC if you are running a > reasonable recent version of UCCX, which shows maximum seats used per time > period. Its under the stock reports Inbound -> System > > > > On Tue, Nov 29, 2016 at 9:29 AM, Ben Amick wrote: > > So realistically, I’m way overlicensed in my environment then? Right now > we have a 1:1 relationship between agents and licenses for both QM and CCX, > with agents working across 2 shifts. > > Is there any report I can run to see how many concurrent users I have at > peak hours? > > > > *Ben Amick* > > Telecom Analyst > > > > *From:* James Buchanan [mailto:james.buchan...@gmail.com] > *Sent:* Tuesday, November 29, 2016 9:27 AM > *To:* Ben Amick > *Cc:* cisco-voip@puck.nether.net > *Subject:* Re: [cisco-voip] Does CCX enforce licensing? > > > > Hello, > > UCCX enforces agent licenses by concurrently logged in agents. So, if you > have ten licenses and ten agents or supervisors are logged in, the eleventh > login will be rejected. > > Thanks, > > James > > > > On Tue, Nov 29, 2016 at 9:24 AM, Ben Amick wrote: > > I was curious about this, I know QM enforces licensing by not letting you > toggle in users for licensing, and I know CUCM and Unity enforce licensing > though PLM, but does UCCX enforce licensing? We’re coming up quick on our > license cap right now and I’m debating if I should be scrambling to get > licensing or just persist in current state while we’re evaluating upgrade > costs. > > > > *Ben Amick* > > Telecom Analyst > > > > > Confidentiality Note: This message is intended for use only by the > individual or entity to which it is addressed and may contain information > that is privileged, confidential, and exempt from disclosure under > applicable law. If the reader of this message is not the intended recipient > or the employee or agent responsible for delivering the message to the > intended recipient, you are hereby notified that any dissemination, > distribution or copying of this communication is strictly prohibited. If > you have received this communication in error, please contact the sender > immediately and destroy the material in its entirety, whether electronic or > hard copy. Thank you > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > <http://cp.mcafee.com/d/FZsS921J5xWX2pEVsj76XCQkmnSkNMV4QsCQkmnSkNPPX9J55BZVYsY-Urhhsd79EVLuWdPp3lpmawECSHIdzrBPpdJnor6TbCZpKqek6nD-LObbW8VdNDHTbFIzCejo7nKOqemeEyCJtdmXOvaxVZicHs3jq9J4TvAXTLuZXTKrKr9PCJhbcmrIlU6A_zMdMjlS67OFek7qVqlblbCqOmdyeG_jy3tyuMS2oa8yvbCNPVEV76MnWhEwdbtFkJkKpH9oQKCy0eHZ3h092g36y3UeRzrPh0vaTMFa14AroodT2Z> > > > > > Confidentiality Note: This message is intended for use only by the > individual or entity to which it is addressed and may contain information > that is privileged, confidential, and exempt from disclosure under > applicable law. If the reader of this message is not the intended recipient > or the employee or agent responsible for delivering the message to the > intended recipient, you are hereby notified that any dissemination, > distribution or copying of this communication is strictly prohibited. If > you have received this communication in error, please contact the sender > immediately and destroy the material in its entirety, whether electronic or > hard copy. Thank you > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > <http://cp.mcafee.c
Re: [cisco-voip] Does CCX enforce licensing?
There is a license utilization hourly report in CUIC if you are running a reasonable recent version of UCCX, which shows maximum seats used per time period. Its under the stock reports Inbound -> System On Tue, Nov 29, 2016 at 9:29 AM, Ben Amick wrote: > So realistically, I’m way overlicensed in my environment then? Right now > we have a 1:1 relationship between agents and licenses for both QM and CCX, > with agents working across 2 shifts. > > Is there any report I can run to see how many concurrent users I have at > peak hours? > > > > *Ben Amick* > > Telecom Analyst > > > > *From:* James Buchanan [mailto:james.buchan...@gmail.com] > *Sent:* Tuesday, November 29, 2016 9:27 AM > *To:* Ben Amick > *Cc:* cisco-voip@puck.nether.net > *Subject:* Re: [cisco-voip] Does CCX enforce licensing? > > > > Hello, > > UCCX enforces agent licenses by concurrently logged in agents. So, if you > have ten licenses and ten agents or supervisors are logged in, the eleventh > login will be rejected. > > Thanks, > > James > > > > On Tue, Nov 29, 2016 at 9:24 AM, Ben Amick wrote: > > I was curious about this, I know QM enforces licensing by not letting you > toggle in users for licensing, and I know CUCM and Unity enforce licensing > though PLM, but does UCCX enforce licensing? We’re coming up quick on our > license cap right now and I’m debating if I should be scrambling to get > licensing or just persist in current state while we’re evaluating upgrade > costs. > > > > *Ben Amick* > > Telecom Analyst > > > > > Confidentiality Note: This message is intended for use only by the > individual or entity to which it is addressed and may contain information > that is privileged, confidential, and exempt from disclosure under > applicable law. If the reader of this message is not the intended recipient > or the employee or agent responsible for delivering the message to the > intended recipient, you are hereby notified that any dissemination, > distribution or copying of this communication is strictly prohibited. If > you have received this communication in error, please contact the sender > immediately and destroy the material in its entirety, whether electronic or > hard copy. Thank you > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > <http://cp.mcafee.com/d/FZsS921J5xWX2pEVsj76XCQkmnSkNMV4QsCQkmnSkNPPX9J55BZVYsY-Urhhsd79EVLuWdPp3lpmawECSHIdzrBPpdJnor6TbCZpKqek6nD-LObbW8VdNDHTbFIzCejo7nKOqemeEyCJtdmXOvaxVZicHs3jq9J4TvAXTLuZXTKrKr9PCJhbcmrIlU6A_zMdMjlS67OFek7qVqlblbCqOmdyeG_jy3tyuMS2oa8yvbCNPVEV76MnWhEwdbtFkJkKpH9oQKCy0eHZ3h092g36y3UeRzrPh0vaTMFa14AroodT2Z> > > > > Confidentiality Note: This message is intended for use only by the > individual or entity to which it is addressed and may contain information > that is privileged, confidential, and exempt from disclosure under > applicable law. If the reader of this message is not the intended recipient > or the employee or agent responsible for delivering the message to the > intended recipient, you are hereby notified that any dissemination, > distribution or copying of this communication is strictly prohibited. If > you have received this communication in error, please contact the sender > immediately and destroy the material in its entirety, whether electronic or > hard copy. Thank you > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 10.6 Historical Reports off by 1hr - No NTP issues -Daylight Savings??
Grace, I had a similar issue one time, was an odd thing when we vmotioned the primary node and ended up having issues in reporting with time/timezone and also issues with database replication on our HA node. Try restarting the CUIC services perhaps? On Fri, Nov 4, 2016 at 12:40 PM, Max Harmony wrote: > All, > > The last 3 days of HR reports has been 1hr off, so for example Agents > logged out at 4:30PM, the reports shows them as logged out at 3:30PM. > > I checked all the UCCX CLI logs and noticed that I am not getting NTP > issues. > The UCCX NTP is the CUCM Publisher, I checked that as well and time was > correct > > Anyone running into this issue, the call center supervisors are freaking > out, I am still trying to investigate the NTPs that CUCM is pointed to, or > the ESXi host machine maybe off. > > Welcome your advice > > Thanks in advance > Grace > > > > -- > -- > Grace Maximuangu > > CloudPOP/InvictaCloud > www.cloudpop.com > > *“Go beyond your limits, push yourself, be the best you can be.* > *Experience new cultures, broaden your horizons, stay connected.”* > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] testing voip list activity
now that there is call forward all on non-primary lines buttons i just sit around with nothing to do anymore ;) On Wed, Oct 19, 2016 at 4:11 PM, James Buchanan wrote: > Ever since Cisco released that fix that corrected every outstanding bug > and added every feature request, we'd had little to talk about. > > On Wed, Oct 19, 2016 at 4:08 PM, Lelio Fulgenzi wrote: > >> >> hmmm no messages for three days, gotta get my fix. >> >> >> --- >> Lelio Fulgenzi, B.A. >> Senior Analyst, Network Infrastructure >> Computing and Communications Services (CCS) >> University of Guelph >> >> 519-824-4120 Ext 56354 >> le...@uoguelph.ca >> www.uoguelph.ca/ccs >> Room 037, Animal Science and Nutrition Building >> Guelph, Ontario, N1G 2W1 >> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > ___________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX CUIC Lost Data
Thanks Brian, that helped. We ended up having to stop replication and re-create it to get it working again. On Wed, Sep 28, 2016 at 11:24 AM, Brian Meade wrote: > Try "show uccx dbreplication tables" then look at the contactcalldetail > table: > > REPLICATE:template_db_cra_g_uccx106sub_uccx_1_3_contactcalldetail > STATE:Active ON:g_uccx106pub_uccx > CONFLICT: Timestamp > FREQUENCY:immediate > QUEUE SIZE: 0 > PARTICIPANT: db_cra:informix.contactcalldetail > OPTIONS: transaction,ris,ats,fullrow > REPLID: 131076 / 0x20004 > REPLMODE: PRIMARY ON:g_uccx106pub_uccx > APPLY-AS: INFORMIX ON:g_uccx106pub_uccx > REPLTYPE: Master > > > On Wed, Sep 28, 2016 at 7:58 AM, Ed Leatherman > wrote: > >> I'm troubleshooting UCCX/CUIC issue with lost data (and assorted weird >> things), and I ran across this gem of a thread that helped me get a >> fingerhold on the issue... >> >> I ran Brian's query "select count(*) from contactcalldetail blahblah for >> yesterday. Primary node has records, secondary has 0. I rebooted both last >> night, and no magic this morning so I'm assuming still a replication >> problem... is there a special set of commands for CUIC data replication? My >> google-fu is failing me on this one. The normal utils dbreplication >> runtimestate reports that everything is happy but i have a suspicion thats >> just for the other ccx datastores. >> >> Issue was triggered by vmotioning primary node off of a host that needed >> repair... really strange issue as it also made the CUIC console think it >> was October already (including all the scheduled reports) until we reset >> that service. system clock was fine though. >> >> I have a TAC case open also but engineer doesn't start till later so i'm >> just trying to get a leg up on the issue before the townsfolk show up at my >> lair with torches and pitchforks. >> >> >> >> >> On Thu, Jan 28, 2016 at 5:39 PM, Brian Meade wrote: >> >>> So overnight this fixed itself and now all of the data is showing in >>> CUIC. There must be some sort of process that runs at night to fix these >>> kinds of things. >>> >>> On Wed, Jan 27, 2016 at 5:48 PM, Brian Meade wrote: >>> >>>> Hey everyone, >>>> >>>> I'm seeing an issue with UCCX 10.6(1)1.39 where database >>>> replication between the 2 UCCX nodes was having issues resulting in no new >>>> call data being populated in the CUIC Historical Reports such as the >>>> Detailed Call by Call CCDR report. >>>> >>>> We fixed the replication issue and now new calls are showing in the >>>> CUIC report successfully but we still have a gap of missing data in the >>>> reports. >>>> >>>> The actual call records look to be in the database: >>>> >>>> admin:run uccx sql db_cra select count(*) from contactcalldetail where >>>> startDateTime >= '2016-01-27 00:00:01' >>>> (COUNT(*)) >>>> -- >>>> 20519 >>>> >>>> But CUIC is only pulling in the new calls. Any ideas how to fix this? >>>> >>>> Thanks, >>>> >>>> Brian >>>> >>> >>> >>> ___ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >> >> >> -- >> Ed Leatherman >> > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX CUIC Lost Data
I'm troubleshooting UCCX/CUIC issue with lost data (and assorted weird things), and I ran across this gem of a thread that helped me get a fingerhold on the issue... I ran Brian's query "select count(*) from contactcalldetail blahblah for yesterday. Primary node has records, secondary has 0. I rebooted both last night, and no magic this morning so I'm assuming still a replication problem... is there a special set of commands for CUIC data replication? My google-fu is failing me on this one. The normal utils dbreplication runtimestate reports that everything is happy but i have a suspicion thats just for the other ccx datastores. Issue was triggered by vmotioning primary node off of a host that needed repair... really strange issue as it also made the CUIC console think it was October already (including all the scheduled reports) until we reset that service. system clock was fine though. I have a TAC case open also but engineer doesn't start till later so i'm just trying to get a leg up on the issue before the townsfolk show up at my lair with torches and pitchforks. On Thu, Jan 28, 2016 at 5:39 PM, Brian Meade wrote: > So overnight this fixed itself and now all of the data is showing in > CUIC. There must be some sort of process that runs at night to fix these > kinds of things. > > On Wed, Jan 27, 2016 at 5:48 PM, Brian Meade wrote: > >> Hey everyone, >> >> I'm seeing an issue with UCCX 10.6(1)1.39 where database replication >> between the 2 UCCX nodes was having issues resulting in no new call data >> being populated in the CUIC Historical Reports such as the Detailed Call by >> Call CCDR report. >> >> We fixed the replication issue and now new calls are showing in the CUIC >> report successfully but we still have a gap of missing data in the reports. >> >> The actual call records look to be in the database: >> >> admin:run uccx sql db_cra select count(*) from contactcalldetail where >> startDateTime >= '2016-01-27 00:00:01' >> (COUNT(*)) >> -- >> 20519 >> >> But CUIC is only pulling in the new calls. Any ideas how to fix this? >> >> Thanks, >> >> Brian >> > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Expressways and second call back???
There is a defect/enhancement request around it also: CSCut26355 Thanks for the tip about whitelisting the recording server Patrick On Thu, Aug 11, 2016 at 6:31 AM, Patrick Robitaille < patrick.robitai...@aqr.com> wrote: > We had the same issue and it's related to call recording. You can > whitelist the recorder servers in the Expressway (Core I think), and > recording will work sometimes but not always. If you try this, you need to > enter separate entries for both the hostname and IP of each Recorder Server. > > Officially recording via MRA isn't supported yet. Fix for us was to > disable recording config for just the line appearance on MRA Phone and > Mobile Jabber - TCT, BOT, TAB. Internal desktop Jabber should work - CSF. > > > Patrick Robitaille > AQR Capital Management > O: (203) 742-3797 | C: (203) 914-9572 > > > On Aug 8, 2016, at 4:25 PM, Ed Leatherman ealeather...@gmail.com>> wrote: > > Scott, > > Are you doing call recording on the line? I ran into a very very similar > thing setting up MRA this weekend, it turned out that my DN was setup for > call recording, and the mystery call was a SIP INVITE from CUCM trying to > get jabber to send media to the call recorder. I don't have it fixed yet > (other than to disable recording) > > On Mon, Aug 8, 2016 at 12:04 PM, Matthew Loraditch heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>> wrote: > EM is technically not supported so I’d want to know if that happens on a > phone that is hardcoded with those dns and such. > > Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA > Network Engineer > Direct Voice: 443.541.1518 > > Facebook<https://www.facebook.com/heliontech?ref=hl> | Twitter< > https://twitter.com/HelionTech> | LinkedIn<https://www.linkedin. > com/company/helion-technologies?trk=top_nav_home> | G+< > https://plus.google.com/+Heliontechnologies/posts> > > From: cisco-voip [mailto:cisco-voip-bounces@http://puck.nether.net cisco-voip-boun...@puck.nether.net>] On Behalf Of Scott Voll > Sent: Monday, August 8, 2016 11:57 AM > To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> > Subject: [cisco-voip] Expressways and second call back??? > > so we are just starting our pilot / testing of Expressways. > > we have a 8861 phone. It authenticates to Expressways. we then log into > extension Mobility. if we log in with a simple DN User it works fine. > > if we log in with one of our profiles, that have Jabber, SNR, different > device profiles, etc we end up with a second call, calling in just after > the call connects or receiving a call just after it connects. soon as we > try to answer it, it goes away. go back to the original call and it comes > back and just keeps ringing. > > Anyone seen anything like this? CM 10.5. expressways 8.7 > > TIA > > Scott > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> > https://puck.nether.net/mailman/listinfo/cisco-voip > > > > > -- > Ed Leatherman > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> > https://puck.nether.net/mailman/listinfo/cisco-voip > > > > > > Disclaimer: This e-mail may contain confidential and/or privileged > information. If you are not the intended recipient or have received this > e-mail in error, please notify the sender immediately and destroy/delete > this e-mail. You are hereby notified that any unauthorized copying, > disclosure or distribution of the material in this e-mail is strictly > prohibited. > > This communication is for informational purposes only. It is not intended > as an offer or solicitation for the purchase or sale of any financial > instrument or as an official confirmation of any transaction. All > information contained in this communication is not warranted as to > completeness or accuracy and is subject to change without notice. Any > comments or statements made in this communication do not necessarily > reflect those of AQR Capital Management, LLC and its affiliates. > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Expressways and second call back???
Scott, Are you doing call recording on the line? I ran into a very very similar thing setting up MRA this weekend, it turned out that my DN was setup for call recording, and the mystery call was a SIP INVITE from CUCM trying to get jabber to send media to the call recorder. I don't have it fixed yet (other than to disable recording) On Mon, Aug 8, 2016 at 12:04 PM, Matthew Loraditch < mloradi...@heliontechnologies.com> wrote: > EM is technically not supported so I’d want to know if that happens on a > phone that is hardcoded with those dns and such. > > > > Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA > Network Engineer > Direct Voice: 443.541.1518 > > Facebook <https://www.facebook.com/heliontech?ref=hl> | Twitter > <https://twitter.com/HelionTech> | LinkedIn > <https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> | > G+ <https://plus.google.com/+Heliontechnologies/posts> > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Scott Voll > *Sent:* Monday, August 8, 2016 11:57 AM > *To:* cisco-voip@puck.nether.net > *Subject:* [cisco-voip] Expressways and second call back??? > > > > so we are just starting our pilot / testing of Expressways. > > > > we have a 8861 phone. It authenticates to Expressways. we then log into > extension Mobility. if we log in with a simple DN User it works fine. > > > > if we log in with one of our profiles, that have Jabber, SNR, different > device profiles, etc we end up with a second call, calling in just after > the call connects or receiving a call just after it connects. soon as we > try to answer it, it goes away. go back to the original call and it comes > back and just keeps ringing. > > > > Anyone seen anything like this? CM 10.5. expressways 8.7 > > > > TIA > > > > Scott > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DTMF interworking on CUBE - asymmetric payloads
Thanks Justin. My concern is with nte's coming in a mix of different payload types depending on the end point. Since it seems I can only specify one payload type for them, when I get an offer using a different value, CUBE doesn't include it in the answer. And then no interworking happens since we didnt' successfully negotiate a DTMF relay method at all. I'm hoping with the asymmteric payload feature that cube will negotiate and pass through the dynamic payload type and also actually look at the packets and interwork them to kpml. The more I think about it the less likely it sounds, but I may as well try it. I think what I really need for it to do is just negotiate EITHER 101 or 98 for dtmf relay out to the 3rd party, but i'm stumped on if/how to make that work. On Mon, Jul 18, 2016 at 9:21 PM, Justin Steinberg wrote: > yes, CUBE can do RFC2833/NTP to a Telco and SIP-KPML to CUCM. I do this > for calls that terminate on CCX IVR since CCX does not support RFC2833. > With only rtp-nte on the dialpeer from CUBE to CUCM, CUCM will invoke a > MTP. Adding sip-kpml to the dial-peer will allow RTP directly from CUBE > to CCX without any MTP in the middle. > > On Mon, Jul 18, 2016 at 5:08 PM, Ed Leatherman > wrote: > >> Thanks Daniel, that helps a lot in understanding the feature. I'm curious >> if CUBE will also translate digits to KPML in this case if the leg to CUCM >> has that negotiated. Wish I had a lab built out for this :) >> >> >> >> On Mon, Jul 18, 2016 at 4:22 PM, Daniel Pagan wrote: >> >>> Ed: >>> >>> >>> >>> I specifically worked with the dynamic payload option for a few cases >>> that came my way. Based on my findings, when a dynamic payload type (such >>> as 100/101/etc.) is received by CUBE, it will check if the next-hop >>> dial-peer has the asymmetric payload feature enabled and, if it is, will >>> pass the received payload type through to the next call-leg. Take a look at >>> my screen shot below. This was taken from some old notes where AT&T was the >>> customer’s carrier. >>> >>> >>> >>> >>> >>> The call flow above shows two call-legs, and *the arrows represent an >>> offer/answer exchange*. >>> >>> >>> >>> With asymmetric payload enabled on both call legs, the 100 offer from >>> ATT is passed to CUCM despite 101 being the default PT for NTE. In the SDP >>> answer from CUCM, we’re getting PT 101 -- since asymmetry is enabled on the >>> DP to ATT in this call flow, we pass the 101 through to ATT despite having >>> received PT 100. >>> >>> >>> >>> This results in asymmetry on our negotiated PT for each call-leg. >>> >>> >>> >>> *Let’s change it up a bit… A second example.* >>> >>> If asymmetry was disabled on the dial-peer to CUCM but enabled to ATT, >>> we would receive 100 PT from ATT, send 101 to CUCM, receive 101 from CUCM, >>> and send 101 to ATT. The resulting PTs would be symmetrical between CUBE >>> and CUCM, but asymmetrical between CUBE and ATT. >>> >>> >>> >>> See screenshot below for a third example: >>> >>> >>> >>> >>> >>> This example shows asymmetric payload disabled on both call-legs using >>> the same call flow. CUBE receives PT of 100 from ATT -- the outbound >>> dialpeer has asymmetry disabled, so it transmits the PT specified for that >>> dial-peer (default 101 or any hardcoded dynamic PT) to CUCM. We then >>> receive 101 from CUCM and, since our inbound dial-peer has asymmetry >>> disabled, CUBE sends 100 to match the original PT it received. Asymmetry is >>> disabled so CUBE is not passing the received dynamic PT through to the >>> next-hop dial-peer - we have symmetry on both call legs for our NTE PT. >>> >>> >>> >>> Note that CUBE has no issues receiving one dynamic PT for NTE and >>> sending another (ex: receiving PT 100 and transmitting 101 for RTP-NTE) on >>> the same call leg. >>> >>> >>> >>> Hope this helps >>> >>> >>> >>> - Dan >>> >>> end attach- >>> >>> >>> >>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On >>> Behalf Of *Ed Leatherman >>> *Sent:* Monday, July 18, 2016 3:10 PM >>> *To:* Cisco VOIP >>> *Subject:* [cisco-voip] DTMF interworking on CUBE - asymmetric payloads >>> >>> &g
Re: [cisco-voip] DTMF interworking on CUBE - asymmetric payloads
Thanks Daniel, that helps a lot in understanding the feature. I'm curious if CUBE will also translate digits to KPML in this case if the leg to CUCM has that negotiated. Wish I had a lab built out for this :) On Mon, Jul 18, 2016 at 4:22 PM, Daniel Pagan wrote: > Ed: > > > > I specifically worked with the dynamic payload option for a few cases that > came my way. Based on my findings, when a dynamic payload type (such as > 100/101/etc.) is received by CUBE, it will check if the next-hop dial-peer > has the asymmetric payload feature enabled and, if it is, will pass the > received payload type through to the next call-leg. Take a look at my > screen shot below. This was taken from some old notes where AT&T was the > customer’s carrier. > > > > > > The call flow above shows two call-legs, and *the arrows represent an > offer/answer exchange*. > > > > With asymmetric payload enabled on both call legs, the 100 offer from ATT > is passed to CUCM despite 101 being the default PT for NTE. In the SDP > answer from CUCM, we’re getting PT 101 -- since asymmetry is enabled on the > DP to ATT in this call flow, we pass the 101 through to ATT despite having > received PT 100. > > > > This results in asymmetry on our negotiated PT for each call-leg. > > > > *Let’s change it up a bit… A second example.* > > If asymmetry was disabled on the dial-peer to CUCM but enabled to ATT, we > would receive 100 PT from ATT, send 101 to CUCM, receive 101 from CUCM, and > send 101 to ATT. The resulting PTs would be symmetrical between CUBE and > CUCM, but asymmetrical between CUBE and ATT. > > > > See screenshot below for a third example: > > > > > > This example shows asymmetric payload disabled on both call-legs using the > same call flow. CUBE receives PT of 100 from ATT -- the outbound dialpeer > has asymmetry disabled, so it transmits the PT specified for that dial-peer > (default 101 or any hardcoded dynamic PT) to CUCM. We then receive 101 from > CUCM and, since our inbound dial-peer has asymmetry disabled, CUBE sends > 100 to match the original PT it received. Asymmetry is disabled so CUBE is > not passing the received dynamic PT through to the next-hop dial-peer - we > have symmetry on both call legs for our NTE PT. > > > > Note that CUBE has no issues receiving one dynamic PT for NTE and sending > another (ex: receiving PT 100 and transmitting 101 for RTP-NTE) on the same > call leg. > > > > Hope this helps > > > > - Dan > > end attach- > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Ed Leatherman > *Sent:* Monday, July 18, 2016 3:10 PM > *To:* Cisco VOIP > *Subject:* [cisco-voip] DTMF interworking on CUBE - asymmetric payloads > > > > I'm trying to get my head wrapped around some DTMF interworking > features... > > > > I have this setup: > > > > UCM -- CUBE --- 3rd party system > > > > For both call legs through CUBE I'm advertising kpml and rtp-nte for > dtmf-relay > > > > The 3rd party sometimes sends me rtp payload type 101 for nte's, and no > kpml, and things work (as a bonus I observed CUBE correctly interworking > the nte's from the pbx into KPML, so uccx didn't break). > > Sometimes they send type 98 and no kpml, and things don't work. > > > > I'm trying to understand what is happening and what feature should fix it > (without breaking other things) > > > > Assumption: > > "dtmf-relay rtp-nte kpml" is telling CUBE to offer/accept rtp type 101 > only for nte. I observe that CUBE negotiates KPML only for the associated > call leg back to UCM and doesn't bother with rtp-nte, so its just like any > other codec that CUBE doesn't care about. > > > > So.. if third party system ONLY sent me dtmf-relay with payload type 98, > could I just set the rtp payload type for this to 98 on the inbound dial > peer? would CUBE then correctly switch these up to 101 headed back to UCM? > > > > How can I (or can I at all) make this work in my particular case were I > could receive both? > > I see "asymmetric payload dtmf" thrown about as a possible solution, but > I'm having trouble understanding what it actually does. It sounds like it > passes these payload types through CUBE, so UCM could be getting rtp > payload type 98 - it knows what to do with it? It seems like then CUBE > wouldn't be able to translate things to KPML this way... > > > > I'm reading > http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-dymc-payld-dtmf.html > but I guess I'm just not drinking enough coffee today (or too much) and I'm > not getting what exactly this command does. > > > > Anyone know how that asymmeteric command works? > > > > -- > > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] DTMF interworking on CUBE - asymmetric payloads
I'm trying to get my head wrapped around some DTMF interworking features... I have this setup: UCM -- CUBE --- 3rd party system For both call legs through CUBE I'm advertising kpml and rtp-nte for dtmf-relay The 3rd party sometimes sends me rtp payload type 101 for nte's, and no kpml, and things work (as a bonus I observed CUBE correctly interworking the nte's from the pbx into KPML, so uccx didn't break). Sometimes they send type 98 and no kpml, and things don't work. I'm trying to understand what is happening and what feature should fix it (without breaking other things) Assumption: "dtmf-relay rtp-nte kpml" is telling CUBE to offer/accept rtp type 101 only for nte. I observe that CUBE negotiates KPML only for the associated call leg back to UCM and doesn't bother with rtp-nte, so its just like any other codec that CUBE doesn't care about. So.. if third party system ONLY sent me dtmf-relay with payload type 98, could I just set the rtp payload type for this to 98 on the inbound dial peer? would CUBE then correctly switch these up to 101 headed back to UCM? How can I (or can I at all) make this work in my particular case were I could receive both? I see "asymmetric payload dtmf" thrown about as a possible solution, but I'm having trouble understanding what it actually does. It sounds like it passes these payload types through CUBE, so UCM could be getting rtp payload type 98 - it knows what to do with it? It seems like then CUBE wouldn't be able to translate things to KPML this way... I'm reading http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-dymc-payld-dtmf.html but I guess I'm just not drinking enough coffee today (or too much) and I'm not getting what exactly this command does. Anyone know how that asymmeteric command works? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] feature unavailable?
Does it do that right away or after setting the number? All phones or just that model or just that one set? I haven't had trouble with "forward all" since the first (and last) time I upgraded to a dot "oh no" release (4.0) On Thu, Jul 14, 2016 at 3:04 PM, Scott Voll wrote: > Anyone seen feature is unavailable when trying to forward all calls? > > 8861 on CM 10.5.2 > > TIA > > Scott > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Disable SUBSCRIBE messages on SIP trunk?
I wonder if you could do this with a normalization script to just drop subscribe messages? I know there is a knob in the sip trunk security profile to have cucm not accept a subscribe, not sure about preventing it in the outgoing direction. would this cause a problem with DTMF relay if CUCM needs to subscribe to kpml from your IVR? I'm thinking of possibility of your IVR wanting to outpulsing DTMF digits for some purpose. On Tue, Jul 12, 2016 at 4:24 PM, Biffle, Gerrad < gerrad.bif...@greensboro-nc.gov> wrote: > Does anyone know if it’s possible to disable the SUBSCRIBE messages on a > SIP trunk from my CUCM to a third party IVR system? If so, any concerns in > doing so? > > > > Thanks for the help! > > === > Please note that email sent to and from this address is subject > to the North Carolina Public Records Law and may be disclosed to third > parties. > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SIP protocol question
Hi Daniel, That was how I read it as well - i just couldn't find an explanation of what the telephone-events package was actually for. Thanks for confirming that the rtp-nte is negotiated fine w/o it. The issue that prompted me to start digging into this was that calls to the med center's help desk call center from certain phones (so far just 7960's) have not been able to connect to an agent. They get IVR audio but when the transfer happens it dies. So I am looking at a trace just from that. It is wild how many re-invites are happening on the med center side that CUBE just consumes. note; I have midcall passthru media-change enabled, nothing else fancy. codec is locked in at g711u, and my dial peers have both kpml and rtp-nte enabled. I'm seeing 3 things that are odd to me... #1 at one point during the dialogue, there is a reinvite from med center that gets passed through cube, but I can't tell why as both CUCM and the med center sbc are continuing to advertise G711ulaw and rtp-nte. #2 on the reinvite that gets passed through, it comes in as a DO from CUBE, and CUCM is sending allow-events: presence,kpml (no telephone-events) along with its normal SDP offer including g711u and rtp-nte in the 200 OK. It's sending a=sendonly, #3 CUBE ack's back to CUCM with a new SDP that includes NO rtp-nte for some reason. In the meantime, it has also advertised SDP back outside to the med center a similar SDP without rtp-nte and a=sendonly. So it removed rtp-nte and passed the sendonly back out to med center At this point, CUCM does its own reinvite to establish sendrcv, however this does not get passed through and a few seconds later I get a BYE from the med center sbc. I did find a bug that was suspiciously similar, CSCsw85869 however i'm not running T.38. I'm going to try and reproduce the issue without all the extra reinvites and maybe check with TAC on it. The trace i'm working from is hideous. CUBE has some MTP resources available but doesn't appear to be inserting them at any point. My next point of investigation is to see if it's unable to for some reason. Since we've only reproduced the issue with a 7960 I didnt want to rule out something weird like that. On Fri, Jul 8, 2016 at 10:35 AM, Daniel Pagan wrote: > According to the RFC, the Allow-Events header is specifically used to > convey events that a UA can support using the SUBSCRIBE method. Typically > we’ll see KPML or Presence as a supported Allow-Event value, which makes > sense since both of those events are initiated through SUBSCRIBE/NOTIFY > transactions. I’m looking through sets of trace files where I know RTP-NTE > is negotiated and have found multiple instances where DTMF was negotiated > successfully (to RTP-NTE) without including telephone-events in the > Allow-Events header. > > > > What type of issues are you seeing with ReINVITEs? > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Ed Leatherman > *Sent:* Friday, July 08, 2016 9:18 AM > *To:* Cisco VOIP > *Subject:* [cisco-voip] SIP protocol question > > > > Happy Friday! > > > > This is probably a dumb basic question but my google-fu is failing me. > > > > What exactly is the "telephone-event" event package used for in the > allow-events: header? Must it be present? > > > > I have been assuming it had to do with rtp-nte but that seems wrong since > that's negotiated in the SDP. > > > > I'm trying to integrate with our associated Medical Center with a SIP > Trunk out of CUBE (IOS 15.4(3)M3) - they are running a mish-mash of Siemens > PBX/SIP/Call Center stuff on their end. I'm having some weird stuff happen > with reinvites when their system(s) transfer the call around and I'm trying > to nail down everything I find that i'm not sure of... in one of the > transactions the telephone-event is missing out of the allow-events in the > invite. > > > > Thanks! > > > > > > > > > -- > > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] SIP protocol question
Happy Friday! This is probably a dumb basic question but my google-fu is failing me. What exactly is the "telephone-event" event package used for in the allow-events: header? Must it be present? I have been assuming it had to do with rtp-nte but that seems wrong since that's negotiated in the SDP. I'm trying to integrate with our associated Medical Center with a SIP Trunk out of CUBE (IOS 15.4(3)M3) - they are running a mish-mash of Siemens PBX/SIP/Call Center stuff on their end. I'm having some weird stuff happen with reinvites when their system(s) transfer the call around and I'm trying to nail down everything I find that i'm not sure of... in one of the transactions the telephone-event is missing out of the allow-events in the invite. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUBE and LTI for MTPs
I've been beating my head on this all day, glad I ran across this thread as it at least seems to rule out one potential issue on my current SIP puzzle :) I've found 2 different documents with tables that suggest KPML <> rpt-nte shouldn't work... i realize its over a year since this thread popped in - has anyone ever seen something to the contrary in official documentation? On Tue, Mar 31, 2015 at 11:54 AM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > After reading a bit more in the CUCM SRND, I see that SIP INFO is not > supported by CUCM, and furthermore, Cisco's preferred OOB method is KPML. > > "Out-of-band (OOB) SIP DTMF signalling methods include Unsolicited Notify > (UN), Information > (INFO), and Key Press Mark-up Language (KPML). KPML (RFC 4730) is the OOB > signalling method > preferred by Cisco and is supported by Cisco Unified CM, Cisco IOS > platforms (Release 12.4 and later), > and most models of Cisco Unified IP Phones. INFO is not supported by > Unified CM." > Source: > http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/trunks.html#pgfId-1346554 > > With that said, according to the link you provided on CUBE and DTMF > inter-working, it would appear as though the only option which remains is > SIP NOTIFY. This method requires the SIP Trunk Security Profile to have > the Accept unsolicited notification checkbox checked. > > I did test with KPML, and CUBE did inter-work it. My IOS/CUBE version is > as follows: > > *CUBE#sh cube status | in Version* > *CUBE-Version : 10.0.2* > *SW-Version : 15.4.3.M2, Platform CISCO2921/K9* > > I think I'll go with KPML for now, and thank you for helping me to see the > light. > > On Mon, Mar 30, 2015 at 1:52 PM Brian Meade wrote: > >> This chart has all the interoperability that can be handled by dtmf-relay >> natively on CUBE- >> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html#concept_264617919921874995299551391601561 >> >> Brian >> >> On Mon, Mar 30, 2015 at 2:29 PM, Brian Meade wrote: >> >>> dtmf-relay I believe should handle that find for you without the MTP. >>> >>> On Mon, Mar 30, 2015 at 1:21 PM, Anthony Holloway < >>> avholloway+cisco-v...@gmail.com> wrote: >>> >>>> Oooo, good question Brian. It's my understanding that in order for the >>>> below specific call flow to work, an MTP is required for DTMF inter-working >>>> of inband to out-of-band. >>>> >>>> PSTN Caller Pushes DTMF ---> ITSP Delivers RFC2833 ---> CUBE Delivers >>>> OOB ---> CUCM Devlier OOB ---> UCCX CTI Port Receives OOB >>>> >>>> Is that not the case? >>>> >>>> On Mon, Mar 30, 2015 at 12:01 PM Brian Meade wrote: >>>> >>>>> What are you trying to accomplish with the MTP that can't already be >>>>> accomplished with media flow-through and dtmf-relay? >>>>> >>>>> On Mon, Mar 30, 2015 at 12:38 PM, Anthony Holloway < >>>>> avholloway+cisco-v...@gmail.com> wrote: >>>>> >>>>>> All, >>>>>> >>>>>> I know the name itself, LTI, includes the word transcoding, but I'm >>>>>> just double checking that this will or will not work for registering an >>>>>> MTP >>>>>> on the CUBE. All roads are leading me to the answer, but it just seems >>>>>> like a huge miss on Cisco's part to not allow us to register MTPs as well >>>>>> as XCODE via the LTI method. >>>>>> >>>>>> This works for me: >>>>>> dspfarm profile 1 transcode >>>>>> codec g711ulaw >>>>>> codec g729ar8 >>>>>> max sessions 1 >>>>>> assoc app cube >>>>>> no shut >>>>>> ! >>>>>> >>>>>> This does not work for me (it hangs on associating to cube app): >>>>>> dapfarm profile 2 mtp >>>>>> codec g711ulaw >>>>>> max sessions software 1 >>>>>> assoc app cube >>>>>> no shut >>>>>> ! >>>>>> >>>>>> I have the required dspfarm and mode border-element commands, and >>>>>> rebooted after as well. >>>>>> >>>>>> Seems like with the standard requirement of rfc2833 on SIP trunks to >>>>>> the ITPS, and
Re: [cisco-voip] Porting numbers between carriers
Circling back for posterity - SIP provider actually had not notified the LEC that the numbers had been setup, so the LEC (imo) legitimately wouldn't drop the numbers out of their routing. I had to badger / escalate with the SIP provider enough to get someone from their porting group on the phone. On Fri, Jun 24, 2016 at 10:46 AM, Adam wrote: > This is like when someone puts a static route in somewhere in the network > and then forgets about it, then wonders why BGP isn't getting them there. I > normally find this happens in scenarios where someone manually configured > something outside of the provider's provisioning tools and it was never > documented. > > -Adam > > > On Friday, June 24, 2016, Ed Leatherman wrote: > >> Good morning, >> >> Is anyone familiar with the process of porting telephone numbers between >> carriers? >> >> I've recently ported 40 numbers from "Carrier A" to "Carrier B". Carrier >> B now has the numbers in their system and most callers to those numbers are >> getting to us via our SIP trunk with Carrier B. >> >> However, callers on local POTS lines with Carrier A are still reaching us >> via our existing PRI with Carrier A. >> >> Carrier B says the porting is complete; Carrier A says B has not >> completed the process and that is why the numbers are still active with >> them. >> >> Aside from me riding Carrier B every day to figure it out, what magic is >> involved behind the scenes here? I miss BGP. >> >> -- >> Ed Leatherman >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Porting numbers between carriers
Thanks Ryan, that makes sense. Carrier A is the normal LEC for the area. On Fri, Jun 24, 2016 at 8:16 AM, Ryan Huff wrote: > Here is a thought Ed > > > I used to work for a CLEC and this was the bane of my existence > getting carriers to admit fault CAN be like shaving a yak while wearing a > meat suit in lion's den ... > > > If Carrier A is the LEC for the area that the POTS lines are in, they may > (likely) still have local routes on their local switches. The net effect is > that when caller local to the area hits these switches, the call is routed > by Carrier A using the old (now stale) route, rather than going out to the > PSTN. > > > In other words, Carrier A may still be treating those POTS numbers as an > ONNET call, for the subscribers it services. If this is the case, I have > dealt with this in two ways; > > > >- Escalate to engineering and explain the issue, in detail >- Have a subscriber of Carrier A call the carrier and say, "when I >dial XYZ number, it takes me to the wrong person". This usually gets the >carrier to trace the call and then they find the route and remove it. > > > = Ryan = > > > > Email: ryanh...@outlook.com > > Spark: ryanh...@outlook.com > > Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> > > LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff> > > Web ryanthomashuff.com > > > -- > *From:* cisco-voip on behalf of Ed > Leatherman > *Sent:* Friday, June 24, 2016 8:03 AM > *To:* Cisco VOIP > *Subject:* [cisco-voip] Porting numbers between carriers > > Good morning, > > Is anyone familiar with the process of porting telephone numbers between > carriers? > > I've recently ported 40 numbers from "Carrier A" to "Carrier B". Carrier B > now has the numbers in their system and most callers to those numbers are > getting to us via our SIP trunk with Carrier B. > > However, callers on local POTS lines with Carrier A are still reaching us > via our existing PRI with Carrier A. > > Carrier B says the porting is complete; Carrier A says B has not completed > the process and that is why the numbers are still active with them. > > Aside from me riding Carrier B every day to figure it out, what magic is > involved behind the scenes here? I miss BGP. > > -- > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Porting numbers between carriers
Good morning, Is anyone familiar with the process of porting telephone numbers between carriers? I've recently ported 40 numbers from "Carrier A" to "Carrier B". Carrier B now has the numbers in their system and most callers to those numbers are getting to us via our SIP trunk with Carrier B. However, callers on local POTS lines with Carrier A are still reaching us via our existing PRI with Carrier A. Carrier B says the porting is complete; Carrier A says B has not completed the process and that is why the numbers are still active with them. Aside from me riding Carrier B every day to figure it out, what magic is involved behind the scenes here? I miss BGP. -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM 11.5 Upgrade from 11.0 Requires PUT?
When I tried to download the 11.5 file a few days ago, CCO showed as "Additional entitlement required" and I couldn't even download it. My SE suggested we 'order' the PUT SW upgrade, which we did yesterday; today I can now download the upgrade file from the same place on CCO. This may just be complete coincidence, just something I noticed. On Thu, Jun 16, 2016 at 2:15 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > I get that the proof is in the pudding, but I personally research and plan > projects based on data on cisco.com. My job requires that the > information on cisco.com is as accurate as possible. Heck, I could do > the upgrade in dCloud, but what's easier: > > A) Spending 5 minutes to look at the upgrade path requirements > B) Spending 6 hours messing around with dCloud and upgrading CUCM 11 to > 11.5 > > Time is money. > > On Thu, Jun 16, 2016 at 12:14 PM, Matthew Loraditch < > mloradi...@heliontechnologies.com> wrote: > >> I’m pretty sure it’s just standard text. Doing a test and will let you >> know. I don’t recall anything in the release notes or partner training >> materials about new licenses. >> >> >> >> Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, CCDA >> Network Engineer >> Direct Voice: 443.541.1518 >> [image: Email_Sig_Template_H_shortcopy_UPDATED] >> >> Facebook <https://www.facebook.com/heliontech?ref=hl> | Twitter >> <https://twitter.com/HelionTech> | LinkedIn >> <https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> >> | G+ <https://plus.google.com/+Heliontechnologies/posts> >> >> >> >> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On >> Behalf Of *Anthony Holloway >> *Sent:* Thursday, June 16, 2016 1:08 PM >> *To:* Cisco VoIP Group >> *Subject:* [cisco-voip] CUCM 11.5 Upgrade from 11.0 Requires PUT? >> >> >> >> Did anyone else see this? >> >> >> >> [image: Inline image 1] >> >> >> >> Is that right? >> > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX Script redirect
I ran across this a few weeks ago and ended up sorting through it the way Anthony describes - that seems like such a strange default behavior to me on the CTI RP. I swear I thought it was NOT like that some time back (UCCX 3 or 4?) because I distinctly remember doing my best to make sure no one could call those CTI ports directly - but never had to deal with that additional setting on the RP. I didn't find it explained all that well either in the docs. On Mon, Jun 13, 2016 at 10:35 AM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > On a related topic of CSS/PT for UCCX that a lot of people miss, is that > you don't need to add your UCCX CTI Port's DN PT to your device CSSs (E..g, > gateways and phones)IFyou set it up right. Here's how... > > Put your CTI Port DNs in a partition labeled something like > UCCX-Private-PT, or UCCX-Restricted-PT, or even > UCCX-Needs-A-New-Script-Editor-PT. The point is, you can name it whatever > you want. > > Then create a CSS for your CTI Route Points, that will house this new UCCX > PT for your CTI Port's DNs. Again, name it whatever you like: > "UCCX-Can-Has-Precision-Routing-Queues-CSS." Now, here's the most > important step, you have to switch the setting "Calling Search Space for > Redirect" on the CTI Route Point from "Default Calling Search Space" to > "Route Point Address Search Space." > > This will allow you to do two things: > > 1) Remove the possibility of someone dialing a CTI Port Directly, which > shouldn't do any harm really, but come on, let's just stop it from even > being possible, because who knows what might happen? > > 2) Use whatever numbering plan and number range you wish, as it will never > conflict with the rest of your Enterprise Numbering Plan or Dial Plan, as > the CTI Route Point will only ever need to reach the CTI Ports, and no > where else. Who here has ever dealt with trying to fit Non-DID DNs into a > +E164 plan and ended up with something like: +1103201 as the CTI Port > DN? > > > Note that this has nothing to do with how you redirect calls off of CTI > Ports as in the original question. That is separate and can be handled in > one of two ways: > > 1) The way Brian mentioned, which is the default, and that is to use the > configured CSS on the CTI ports to route calls > > 2) Change the CTI Port CSS "Redirect Calling Search Space" from "Redirect > Party" (aka the CTI Ports) to "Calling Party" (aka the device which made > the call into UCCX; for PSTN callers this is your gateway/trunk). > > It should be obvious that "Calling Party" only works if there is a calling > party. This will not work if you have an HTTP triggered application which > places an outbound call. Therefore, use your port groups wisely. > > > On Thu, Jun 9, 2016 at 3:59 PM, Brian Meade wrote: > >> It uses the CSS of the CTI Ports for these redirects. I usually put >> these type of specific UCCX redirect route patterns in one of my UCCX >> partitions to fix these kinds of things. >> >> On Thu, Jun 9, 2016 at 4:52 PM, Aaron Banks >> wrote: >> >>> I'm having a little trouble with a script redirect. The redirect works >>> fine, but the call is being sent to a PRI instead of a SIP trunk. How it >>> works is the call comes in over the SIP trunk to a RP and then is sent to >>> UCCX script with a couple of options. Selecting either option sends the >>> call to a DID outside of the enterprise. I have put in a route pattern >>> that is an exact match to one of the options selected in the IVR. My >>> question is - do I have a pattern matching problem or is the CSS of the CTI >>> port being used (which, if true, I know exactly what the problem is). I >>> had the redirect CSS on the trigger set to be the route point, but that >>> clearly is not working. >>> >>> >>> Any comments, suggestions, mild criticism appreciated. >>> >>> >>> Aaron >>> >>> _______ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Old phone models dropped from CUCM 11.5
Interesting. I just started getting a bunch of 7912's boxed up for surplus. I hope we get some more warning when they axe the 7960 model. On Thu, Jun 9, 2016 at 6:55 PM, Stephen Welsh wrote: > Wow, formally dropping phone support from CUCM in 11.5, I assume this > means the models listed will not register... > > > http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/rel_notes/11_5_1/cucm_b_release-notes-cucm-imp-1151/cucm_b_release-notes-cucm-imp-1151_chapter_01.html > > Also, > Very grateful for the MigrationFX ‘plug’ in the CUCM release notes ;) > > FYI: MigrationFX now includes the ability to provision new phones too > (similar to TAPS but no IVR/UCCX, a simple XML service instead), BAT import > the phone details with dummy MAC addresses, use the Migration XML phone > service on the newly plugged in handset (i.e. auto registered with Idle URL > via Universal Device Template). Using the XML service search for the users > extension then MigrationFX will rename the matching BAT device to the MAC > address of the new phone. It will even migrate if the model of the new > phone does not match the BAT import, so more flexibility on deploy handsets > than TAPS. > > We also have some new MigrationFX functionality due for release at Cisco > Live, we shall publish the details in due course, but if you are curious > drop me a line or come and say hello. > > Kind Regards > > Stephen Welsh > CTO > UnifiedFX > > Come along and meet the UnifiedFX team > World of Solutions stand T2 > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCE 11.x and Jabber ?
Hi Jason! No EM on jabber. Not sure about the UCCE question. We are starting to use it from some UCCX (10.6) applications with Finesse. On Mon, Jun 6, 2016 at 1:26 PM, Jason Aarons (AM) < jason.aar...@dimensiondata.com> wrote: > > > Should a customer go with Jabber for use with Finesse/CAD in UCCE 11.x ? > or stick with IP Communicator 8.6.4 ? Is Jabber still Finesse only? > > > > Remind me does Jabber does Extension Mobility login? > > > > Hope all has been well here on Puck. Had a large R&S project last 2 years, > then hiked the AT awhile. > > > > Jason Aarons, CCIEx2 No 38564 > > Consultant > > Dimension Data > > 904-338-3245 mobile > > > > > > > > > > > > > This email and all contents are subject to the following disclaimer: > "http://www.dimensiondata.com/emaildisclaimer"; > <http://www.dimensiondata.com/Global/Policies/Pages/Email-Disclaimer.aspx> > > _______ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] openSSH / SFT / DRS important FYI
Thanks for the heads up Ryan i'm sure i'd have hit this one sooner or later. On Wed, Jun 1, 2016 at 7:10 PM, Ryan Huff wrote: > This is an important FYI for anyone that uses OpenSSH, and by extension > any software that uses OpenSSH. A coworker and I discovered this issue > today by way of using Linux with OpenSSH as a SFTP>DRS target for UC > Manager. > > > Applied to context; in the new OpenSSH 7.2p2, which you'll likely run into > in recent, package managed Linux distributions (Ubuntu, Debian etc) > OpenSSH has disabled weak crypto ciphers by default. Specifically; aes128-cbc, > 3des-cbc,blowfish-cbc (and the use of no cipher) which as of CUCM > 11.0.1.21900-11 are still being used. > > > If you hit this issue: > > > In UC Manager if you try to add a backup device that uses OpenSSH 7.2p2 > you'll get, "unable to access SFTP server. Please check username and > password". Thats because it is failing the key exchange with the OpenSSH > server and getting spanked. > > > On the OpenSSH side, if you look in the output log (in Linux it is > typically /var/log/auth.log) you'll see, "Jun 1 14:06:34 SERVER_HOST > sshd[23578]: fatal: Unable to negotiate with XXX.XXX.XXX.XXX port 33934: no > matching cipher found. Their offer: aes128-cbc,none,3des-cbc,blowfish-cbc > [preauth]". The OpenSSH output is handy because it tells you exactly what > the peer (UC Manager in this case) is looking for. > > > The solution is to add support for 1 or more of these ciphers back into > the OpenSSH server configuration. Typical Linux distributions have this at > /etc/ssh/sshd_config and it looks like, "Ciphers > aes128-cbc,3des-cbc,blowfish-cbc". Just to err on the side of caution I > would add a few of the ciphers that UC Manager is looking for. > > > Hope this saves some pain, > > = Ryan = > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PS-ALI DID Registration
Verizon was web based at least up until they sold their local service to Frontier in my area. I remember the modem based version though. :) On Wed, Jun 1, 2016 at 1:30 PM, Erick Wellnitz wrote: > Intrado makes a product so you can update the locations any time with as > much detail as local law requires. Some other companies offer this as > well, Verizon did at one time and may still offer their PS-ALI serivce but > it was still modem based last time I used it. > > Intrado will integrate directly into your CER so that's a plus if single > point of management is desired. > On Jun 1, 2016 10:26 AM, "Jonathan Charles" wrote: > >> We are setting up CER in Nebraska and the local provider (Cox) is >> claiming they do not support E911... (they won't register the DIDs int he >> PS-ALI database) and advised us to go to Intrado... >> >> What are our options for registering ERLs in the PS-ALI database... >> >> >> >> Jonathan >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > _______ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Jabber for Win / Alerting name
Hi everyone, I had a report from a user this morning that they called someone using jabber for win and the wrong name came up as the alerting name. The DN has the correct alerting name for the person, so I am guessing jabber is doing a lookup somewhere. I found the following doc about calling party name: http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-windows/116433-probsol-jabber-00.html Anyone know if Jabber does the same shenanigans for alerting name? LDAP seems to be correct so if this applies I'm guessing it's something wrong from outlook stuck in jabber's name cache. It's a VP level so I don't have immediate access to just go wipe it out and try it. I can't reproduce the issue calling the same number myself, so it appears to be local to him. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SIP door phone with video camera
Thanks for the suggestions everyone! On Wed, May 18, 2016 at 11:01 AM, Croft, Keith wrote: > Ed, > > I just installed this one and it works great for remote door monitoring > and control. > > You can also configure the device to stream the video feed without the > need to dial the front door. > > > > Part# 2N-9137111CKU 2n Helios Ip 1 Button + Keypad + Camera > > > > CUCM Setup > > https://faq.2n.cz/pages/viewpage.action?pageId=24052239 > > > > > > Best regards, > > *Keith Croft | Collaboration Engineer* > > World Wide Technology, Inc > > > > > > > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *Ed Leatherman > *Sent:* Tuesday, May 17, 2016 8:53 AM > *To:* Cisco VOIP > *Subject:* [cisco-voip] SIP door phone with video camera > > > > Anyone seen a product that does this reasonably well with CUCM 10+? > > > > Use case: Delivery guy shows up, hits button, audio opens up to a video > capable IP phone along with one-way video so the IP Phone can see the guy. > Doesn't need 2 way video. > > > > IP Phone will probably be either 8945 or 9900. > > > > Thanks! > > > > -- > > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CCX 10.6 (HA) CUIC Missing Queue Report after Failover
So if you manually re-run those reports, is the data there? Are both nodes back online or is primary still off line - perhaps the data in question hadn't replicate when the failure occurred. If they are both back online, is the replication OK? On Tue, May 17, 2016 at 12:29 PM, Max Harmony wrote: > >> I need guidance on an issue that I am running into on CUIC. >> >> >> Our UCCX cluster is HA, we failed from P to S and now missing some >> reports that get generated nightly via a data source, specifically the >> ContactQueueDetail table is missing in our reports, can anyone guide me in >> the right direction, I want to check and see if the data exist in UCCX, or >> was actually generated but just not passed over to the datasource >> >> Where do I begin? >> > > > > -- > -- > Grace Maximuangu > > CloudPOP/InvictaCloud > www.cloudpop.com > > *“Go beyond your limits, push yourself, be the best you can be.* > *Experience new cultures, broaden your horizons, stay connected.”* > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] SIP door phone with video camera
Anyone seen a product that does this reasonably well with CUCM 10+? Use case: Delivery guy shows up, hits button, audio opens up to a video capable IP phone along with one-way video so the IP Phone can see the guy. Doesn't need 2 way video. IP Phone will probably be either 8945 or 9900. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] LDAP, Sync, Filters and CUCM
We just ran into the behavior with some agent username changes last week, and it was not even LDAP related - the technician was just trying to be helpful and update some local end user id's for consistency and didn't think that change needed any pre-testing. When he noticed it, he changed all the names back but their skills such were still cleared out. Luckily it was a small number of accounts and we were able to get skills re-assigned quickly, but I think it will still have a minor impact on the agent reporting. On Tue, May 3, 2016 at 4:41 PM, Justin Steinberg wrote: > CUCM doesn't delete the users when they are marked inactive. you can fix > the LDAP agreement, resync and get them back if you get it fixed before the > garbage cycle clears them out. > > this is really an issue for the UCCX team, they should handle user changes > in CUCM more gracefully. This is also the case for situations where the > username changes. If the account changes in the LDAP directory, CUCM will > usually see the change and update the username. however, UCCX doesn't > track the same way and will delete the agent's old account and create a new > account for the new username. the new UCCX account will have no skills, no > team, no associated historical data, etc.I haven't tested this with > UCCX 11, but this is how it has been on UCCX when I last tested it. > > On Tue, May 3, 2016 at 3:04 PM, Anthony Holloway < > avholloway+cisco-v...@gmail.com> wrote: > >> This is related to my post I just made on UCCX and LDAP via CUCM. >> >> I also just found out that a CUCM with an already synced user database >> behaves in the following two ways: >> >>1. If you modify the filter such that it matches 0 records, the sync >>doesn't happen at all. No users are marked as Inactive, no users are >>pulled in, and no users are updated >> >>You will see this in the DirSync log >>Dirsync synched zero users. Please verify the custom LDAP filter >>configured for this agreement >> >>2. However, if you modify the filter such that it matches a single >>record, the sync does happen. All of the non-matched users will become >>Inactive. >> >>You will see this in the DirSync log (the value 1660 will vary by >>scenario) >>DSDBInterface.setUserInactive Found 1660 users in database needing >>update >> >> For #1, it seems like this might be a protection mechanism, preventing >> you from destroying your entire corporate directory. Because, recall that >> EM, Jabber, Finesse, etc., all require your account to be Active Synced in >> order to authenticate you; therefore, making 1660 people go Inactive will >> have a large impact. Or perhaps it was a coding error, and they should >> have made all users go Inactive? >> >> For #2, if we're thinking #1 could be a protective mode, then wouldn't >> 100% user loss be just as bad as 99%? Perhaps the protection mechanism >> should look for a smaller percentage drop in Active users and prohibit an >> LDAP update at that time and display a warning on the page (I.e., Like the >> last known good backup now shows up on the About page). >> >> What do you think? Have you seen this before? Has it bit you? Am I >> missing something obvious? Let me know. Thanks. >> >> _______ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco Voice Training
Are you after a cert or looking for some particular skill set? I'm using videos on ine.com to study for a re-certification, their collab videos seem pretty solid but they are not cisco "authorized" so i can't comment on how much they toe the line. On Mon, May 2, 2016 at 4:04 PM, Aaron Jenkins wrote: > Looking for recommendations on Training Facilities to take Cisco Voice > classes. > > > > Thanks. > > > > > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Additional historical reports available on CCO
Hi Abhiram, Thanks for all the background info, and the heads up about SU2 (i've been waiting on that one for another bug fix) Since the new reports are pre-bundled in SU2, do you foresee any problems applying SU2 if we've added or tried to add one or more of the new reports already? I am mainly thinking of SP's that have been added already. Ed On Mon, Apr 11, 2016 at 9:57 PM, Abhiram Kramadhati (akramadh) < akram...@cisco.com> wrote: > Hi all, > > We had a quick discussion yesterday to address this and here is the > summary: > >1. CUIC currently executes the SP’s as (uccxhrc user) - this is a >problem due to an Informix defect, corrected in the next release. Why is >this a problem? According to Informix, for a user to delete the temp tables >the user has to be the owner of the same. The owner of the temp table is >usually designated to be the same as the owner of the SP – *uccxhruser* >in our case. Now, since the execution is happening as *uccxhrc* user >and the owner is *uccxhruser*, there is a conflict and hence the >errors. Refreshing the CUIC datasource fixes this temporarily because temp >tables are cleared out. The permanent fix is to import the SP as the >*uccxhrc* user. Since the *uccxhrc* user does not have default >privileges to create SP’s, please call into TAC and they can run the >commands for you from the root. This is the defect CSCul06940 >2. We have pre-bundled these reports in 10.6(1)SU2 and the same has >been released yesterday. Since they are pre-bundled, you will not have this >problem. >3. 11.0(1) currently does not have this pre-bundled, since it was >released a few months back. So if you upgrade from 10.6(1)SU2 to 11.0(1), >you will lose these reports – you will have to reimport them. So please be >mindful of point 1. Tentatively, we will pre-bundle them on SU’s on >11.0(1). So you should be able to upgrade to that from 10.6(1)SU2 and still >have the reports. > > > The defect CSCul06940 will affect other custom reports as well (because of > point1). However, in 11.5 the user *uccxhrc* has been deleted. That is > the long term fix. It is an informix fix, hence the challenges to back > port. > > Updated release notes for the reports will be uploaded soon. Please let me > know if you have any questions. > > Regards, > Abhiram Kramadhati > Technical Solutions Manager, CCBU > CCIE Collaboration # 40065 > > From: cisco-voip on behalf of > akramadh > Date: Monday, 11 April 2016 at 10:41 AM > To: Kevin Przybylowski , Ed Leatherman < > ealeather...@gmail.com>, Anthony Holloway , > Justin Steinberg > > Cc: "cisco-voip@puck.nether.net" > Subject: Re: [cisco-voip] Additional historical reports available on CCO > > Hi all, > > Let me find more details on the issue you are facing. We introduced these > reports over the last quarter and more will be available as well; and all > of them have undergone the regular testing cycle. But let me see how best > to address this. > > *Abhiram Kramadhati* > Technical Solutions Manager > Customer Solutions Success team, CCBU > akram...@cisco.com > Phone: *+61 2 8446 6257 <%2B61%202%208446%206257>* > > CCIE Collaboration - 40065 > > *Cisco Systems Australia Pty Limited* > The Forum > 201 Pacific Highway > 2065 > St Leonards > Australia > Cisco.com <http://www.cisco.com/web/AU/> > > <http://wwwin.cisco.com/marketing/corporate/brand/intelbrand/brandstrat/signature/Insert%20your%20LinkedIn%20link> > > Think before you print. > > This email may contain confidential and privileged material for the sole > use of the intended recipient. Any review, use, distribution or disclosure > by others is strictly prohibited. If you are not the intended recipient (or > authorized to receive for the recipient), please contact the sender by > reply email and delete all copies of this message. > > Please click here > <http://www.cisco.com/web/about/doing_business/legal/cri/index.html> for > Company Registration Information. > > From: cisco-voip on behalf of Kevin > Przybylowski > Date: Saturday, 9 April 2016 at 1:45 AM > To: Ed Leatherman , Anthony Holloway < > avholloway+cisco-v...@gmail.com>, Justin Steinberg > Cc: "cisco-voip@puck.nether.net" > Subject: Re: [cisco-voip] Additional historical reports available on CCO > > You also have to put together that Cisco’s official new ccx reports would > fall under the ‘custom’ umbrella! > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net > ] *On Behalf Of *Ed Leatherman > *Sent:* Friday, April 8, 2016 11:37 AM > *To:* Anthony Hollowa
Re: [cisco-voip] Additional historical reports available on CCO
Hours ago actually lol, I never looked through the sp !
On Fri, Apr 8, 2016, 11:20 AM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:
> Defect in question:
> https://bst.cloudapps.cisco.com/bugsearch/bug/CSCul06940
>
> Funny, Ed himself just posted this defect to the list a few days ago. ;)
>
> On Fri, Apr 8, 2016 at 10:16 AM, Justin Steinberg
> wrote:
>
>> those reports have the temp tables issue. they need to be imported
>> using the uccxhrc user.
>>
>> Which makes me really question using these reports in production since
>> the QA on these must have been really minimal to document that they should
>> be imported with uccxhruser.
>>
>> On Fri, Apr 8, 2016 at 11:14 AM, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>>> Update: I rebooted my UCCX to see if that helped, and now I cannot run
>>> the Reason Code Report by Agent Grouping report, as it fails with the same
>>> database error. So, it got worse.
>>>
>>> On Fri, Apr 8, 2016 at 9:44 AM, Ed Leatherman
>>> wrote:
>>>
>>>> Hmm i'll try the other ones - both reason code reports resulted in the
>>>> same errors. I tried the second one using CUIC as the data source just to
>>>> see if it made a difference. Thats the same type of error that i got
>>>> though. Trying to find out if TAC will support these or not also.
>>>>
>>>> On Fri, Apr 8, 2016 at 10:42 AM, Anthony Holloway <
>>>> avholloway+cisco-v...@gmail.com> wrote:
>>>>
>>>>> Update: After importing all reports, I can only run two of the five:
>>>>>
>>>>>1. All Agent Fields Report - GOOD
>>>>>2. Contact Service Queue Activity by Window Duration - Database
>>>>>status Failed
>>>>>3. CSQ All Fields Report - Database status Failed
>>>>>4. Reason Code Report by Agent Grouping - GOOD (This was
>>>>>coincidentally the first one I imported)
>>>>>5. Reason Code Report by Reason Code Grouping - - Database status
>>>>>Failed
>>>>>
>>>>> As one example of the error output from a failed run is:
>>>>>
>>>>> *Error information:*
>>>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDbException:
>>>>> DbException: CuicDataSourceServiceManagerImpl.getDataSet() { Nested
>>>>> SQLException; SQLState: IX000 Vendor code: -313 Message: Not owner of
>>>>> table. } at
>>>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDataSourceServiceManagerImpl.getDataSetBuilder(CuicDataSourceServiceManagerImpl.java:952)
>>>>> at
>>>>> com.cisco.ccbu.cuic.businesslogic.engine.CuicReportEngineWorker.executeQueryUsingDatasetBuilder(CuicReportEngineWorker.java:80)
>>>>> at
>>>>> com.cisco.ccbu.cuic.businesslogic.engine.CuicReportEngineWorker.runReport(CuicReportEngineWorker.java:37)
>>>>> at
>>>>> com.cisco.ccbu.cuic.businesslogic.enginebase.Worker.run(Worker.java:329)
>>>>> at
>>>>> com.cisco.ccbu.infra.threads.InstrumentedRunnable.run(InstrumentedRunnable.java:92)
>>>>> at
>>>>> java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1145)
>>>>> at
>>>>> java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:615)
>>>>> at java.lang.Thread.run(Thread.java:724) at
>>>>> com.cisco.ccbu.infra.threads.ThreadPoolThread.run(ThreadPoolThread.java:164)
>>>>> Caused by: java.sql.SQLException: Not owner of table. at
>>>>> com.informix.jdbc.IfxSqli.a(IfxSqli.java:3643) at
>>>>> com.informix.jdbc.IfxSqli.E(IfxSqli.java:3974) at
>>>>> com.informix.jdbc.IfxSqli.dispatchMsg(IfxSqli.java:2695) at
>>>>> com.informix.jdbc.IfxSqli.receiveMessage(IfxSqli.java:2611) at
>>>>> com.informix.jdbc.IfxSqli.a(IfxSqli.java:1830) at
>>>>> com.informix.jdbc.IfxSqli.executeStatementQuery(IfxSqli.java:1768) at
>>>>> com.informix.jdbc.IfxSqli.executeStatementQuery(IfxSqli.java:1699) at
>>>>> com.informix.jdbc.IfxResultSet.a(IfxResultSet.java:210) at
>>>>> com.informix.jdbc.IfxStatement.executeQueryImpl(IfxStatement.java:1237) at
>>>>> com.informix.jdbc.IfxPreparedStatement.executeQuery(IfxPreparedStatement.java:401)
>>>>> at
>>>>> com.informix.jdbc.IfxCallableStatement.executeQuery(IfxCallableStatement.java:241)
>
Re: [cisco-voip] Additional historical reports available on CCO
CCX_TK_B927C100_00> >>8. Now I'm connected to UCCX DB, and I can see the DB details in AGS >>Server Studio >>9. I expand Databases on the left, then right click the db_cra >>database and select New > SQL Editor >>10. I open the .sql commands in Notepad++ for the first report in the >>file from the ZIP archived CCX_NEWREPORTS\Report Stored >>Procedures\sp_reasoncode_agent_grouping.sql >>11. Copy those commands from Notepad++ and paste them into the SQL >>Editor pane in AGS Server Studio >>12. Click the Execute button (Green Arrow in Tools Bar) Or you can >>press CTRL+E >>13. Scroll through output in bottom pane and check for errors, I saw >>none >>14. Logged into CUIC as application administrator >>15. Created a new folder under the Reports folder and named it >>"Non-Stock" >>16. Clicked Import Reports button at top of Reports page >>17. Clicked Browse to open the report definition for the report I'm >>importing CCX_NEWREPORTS\Report Templates\New Reports\Reason Code Report >> by >>Agent Grouping.zip >>18. Selected the "Non-Stock" folder as the Save To folder and clicked >>Import >>19. Got an error about needing to pick data sources, so I picked UCCX >>as the data source for both the definition and the value lists, then >>clicked Import again >>20. I now see a new folder under the folder "Non-Stock" for the >>report I just imported >>21. Within that new folder, which is the name of the report "Reason >>Code Report by Agent Grouping" the report itself is in there, with the >> same >>name "Reason Code Report by Agent Grouping" (Nesting is good right?) >>22. I click the new report, set some filter, run it and bam, I have a >>new report >> >> The hardest part was obtaining, installing, registering, and configuring >> AGS Server Studio to connect to the UCCX database. With that done, I can >> start on step 9 for the remainder of the reports. >> >> On Fri, Apr 8, 2016 at 8:30 AM, Ed Leatherman >> wrote: >> >>> Good morning, >>> >>> It was pointed out to me that there were a handful of new historical >>> reports for UCCX/CUIC 10.6/11 up for download on CCO in the downloads >>> section. Others might be interested also. >>> >>> Anyone try these out? I added in the Reason Code Report by agent (added >>> SP and template OK) but I just get a Dataset status is Failed (Database >>> error) and then a big pile of java error that makes no sense to me. I'm >>> importing the report as the appadmin user account so that I can import it >>> into the stock reports folder for my internal customers and change >>> permissions on it so they can all access it. >>> Thanks! >>> >>> -- >>> Ed Leatherman >>> >>> ___ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting
Glad to help. I've had tac do the workaround before so if you find you need the temp tables it's a easy fix - they just have to shell into your UCCX to create the extra account for you. On Fri, Apr 8, 2016 at 10:18 AM, Bill Talley wrote: > Closing the loop here, this was my issue. I was able to execute a stored > procedure which wasn't creating a temp table. Thanks again for sharing Ed. > > On Fri, Apr 8, 2016 at 7:27 AM, Ed Leatherman > wrote: > >> Bill, >> >> Does your SP use temporary tables? look at CSCul06940 - I dont know that >> its resolved in 10.X yet. >> >> Ed >> >> >> >> On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley wrote: >> >>> Hi all, >>> >>> Has anyone imported custom stored procedures into the db_cra for use in >>> CUIC? I'm having an issue getting a new stored proc loaded using either >>> RazorSQL or AGS Server Studio. From what I've read, I need to execute the >>> stored proc; however, when I do that I get an error that uccxhruser doesn't >>> have permission to execute the stored proc. It seems I can't apply >>> permissions to the uccxhruser account on the stored proc until the stored >>> proc is loaded. Would anyone have any tips on how I can go about getting >>> the stored proc loaded to the database? >>> >>> Hopefully this makes sense to someone ;-) >>> >>> TIA, >>> Bill >>> >>> ___ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >> >> >> -- >> Ed Leatherman >> > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Additional historical reports available on CCO
Good morning, It was pointed out to me that there were a handful of new historical reports for UCCX/CUIC 10.6/11 up for download on CCO in the downloads section. Others might be interested also. Anyone try these out? I added in the Reason Code Report by agent (added SP and template OK) but I just get a Dataset status is Failed (Database error) and then a big pile of java error that makes no sense to me. I'm importing the report as the appadmin user account so that I can import it into the stock reports folder for my internal customers and change permissions on it so they can all access it. Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting
I haven't touched this in 2 years - here's notes I took when I was playing around with this before in case it will help: http://www.evernote.com/l/AAVCJ8P2UbJG5o0yqjxIO6-9p7edydMHCfE/ On Fri, Apr 8, 2016 at 8:27 AM, Ed Leatherman wrote: > Bill, > > Does your SP use temporary tables? look at CSCul06940 - I dont know that > its resolved in 10.X yet. > > Ed > > > > On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley wrote: > >> Hi all, >> >> Has anyone imported custom stored procedures into the db_cra for use in >> CUIC? I'm having an issue getting a new stored proc loaded using either >> RazorSQL or AGS Server Studio. From what I've read, I need to execute the >> stored proc; however, when I do that I get an error that uccxhruser doesn't >> have permission to execute the stored proc. It seems I can't apply >> permissions to the uccxhruser account on the stored proc until the stored >> proc is loaded. Would anyone have any tips on how I can go about getting >> the stored proc loaded to the database? >> >> Hopefully this makes sense to someone ;-) >> >> TIA, >> Bill >> >> ___________ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > > -- > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting
Bill, Does your SP use temporary tables? look at CSCul06940 - I dont know that its resolved in 10.X yet. Ed On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley wrote: > Hi all, > > Has anyone imported custom stored procedures into the db_cra for use in > CUIC? I'm having an issue getting a new stored proc loaded using either > RazorSQL or AGS Server Studio. From what I've read, I need to execute the > stored proc; however, when I do that I get an error that uccxhruser doesn't > have permission to execute the stored proc. It seems I can't apply > permissions to the uccxhruser account on the stored proc until the stored > proc is loaded. Would anyone have any tips on how I can go about getting > the stored proc loaded to the database? > > Hopefully this makes sense to someone ;-) > > TIA, > Bill > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Present Callers queued time to UCCX Finesse agent desktop
Seems to me like once it gets to the connected branch, its no longer waiting which could account for the "Current" wait duration being -1. Maybe if you change the "Connect" Setting to NO in the Select Resource step, which should change the Connected branch to Selected, then do your calculation, then manually connect? I've never tried this before so its just a guess :) On Thu, Mar 31, 2016 at 4:13 AM, Matthew Collins wrote: > Hi All, > > > > Trying to present the callers queued time to the agent when the agent is > presented the call but not have much luck. > > > > What I have done so far is to get a reporting statistic current Wait > Duration from CSQ IPCC Express and store that as a Integer Call Variable > > > > > > I have played around with where I place the two steps but going around in > circles. In the above example the agents time is queue is showing blank > and when I debug the Time in Queue is set at “-1” > > > > TAI > > > > Matt > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Reporting in inactive endusers in CM
We've go something weird going on with LDAP whereby i'm randomly getting end users marked inactive in cucm (and unity) after a periodic sync. Packet cap shows inconsistent number of search results back from the ldap server (oracle) - so i'm chalking it up to CM doing exactly what its supposed to do. Since I don't manage that ldap service i'm at the mercy of the folks that are, right now they are trying to replicate the issue but its so or miss , haven't been able to yet. In the meantime though, i'd like to try and report on when ldap sync'd users get marked inactive. This way we can keep an eye on it, and if i all the sudden see 1000 users marked inactive, we can manually go in and kick off a new re-sync. or when certain VP's accounts get axed and they can no longer sign into jabber (><) we can proactively go fix it before they have a problem. My first thought is to script something that will do a sql query against the end user table and send back user ids that are inactive - on a schedule so it just shows up in our inboxes in the morning. Anyone ever had to do this, is there a better way? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VMTools - CUCM 10.5
So yeah - we were actually getting ready to log in a change board request to do the reboots to get vmtools installed and then selinux back to enforce, which would have happened while I was away on vacation out of cell phone range :) Soo glad I glanced through my emails today and saw Scott and Baha's messages!!! We're going to get a case started anyway as he has already done the whole operation on PLM. On Thu, Feb 25, 2016 at 7:09 AM, Baha Akman (makman) wrote: > I’m sorry to report, the problem with latest VMTools update failure is not > like CSCul78735 as many of you may have experienced when you upgraded to > CUCM 10.0 and was familiar with the workaround. > > Latest ESXi 5.5 Builds as well as 6.0 Builds bundles a new version of > VMtools version 10.0.0.50046 (build-3000743) aka (10240) > > See https://packages.vmware.com/tools/versions for tools version mapping. > > This new VMTools has a new functionality built-in called vmware-caf See > release notes of it here; > http://pubs.vmware.com/Release_Notes/en/vmwaretools/1000/vmware-tools-1000-release-notes.html > > CUCM 9.1 releases are immune to this new functionality their selinux > policies don't interfere with it. However CUCM 10.X and 11.X builds > explicitly block vmtools-caf from functioning properly. > > If you managed to follow the workaround documented in CSCul78735 where you > put selinux to permissive, update vmtools to this new 10.0 release, then > put it back to enforcing, you will run out of Root Disk Space and Virtual > Memory. > > The new defect tracking this issue is CSCux27503 - "Vmware Tools update on > ESXi 6.0 is failing” please inform TAC about this as the word is still > getting around. Not sure if this is going to be the ultimate defect to fix > it but track this one for now. > > For those of you who have not yet experienced this serious issue, My > suggestion would be to hold off on updating your VMTools after you have > patched your ESXi 5.5 or 6.0 builds to the latest builds. If you have > already put selinux to permissive mode as a workaround to get vmtools > updated DO NOT put it back to enforcing mode. Contact TAC. If you have > already put it back to enforcing mode after upgrading vmtools to 10.0 then > contact TAC immediately as you will certainly run out of Memory and Disk > Space soon. > > Hate to report this here, but this one is a doozy. > > > -- > Baha > > > On Feb 25, 2016, at 6:46 AM, Ed Leatherman wrote: > > We just patched vsphere this past weekend and ran into the selinux bug > with vmtools, and have also noted some weird stuff with memory and disk > space but on Unity Connection. I'll check in with the guy that was handling > it and see if there was any other strangeness, this doesn't give me warm > and fuzzies. > > On Wed, Feb 24, 2016 at 9:13 PM, Hughes, Scott GRE-MG < > shug...@grenergy.com> wrote: > >> Has anyone dealt with bugID >> CSCul78735 lately? >> We just installed our latest round of VMware 5.5 patches. Along with it >> came new VMtools. Usual (automatic) upgrade method did not work. The VMware >> tools status went from 'out of date' to 'not installed' >> >> We were able to get the tools installed on our two subscribers by >> switching SELinux to permissive mode as the bug describes. We switched them >> back to enforcing. Then, we found that the active partition filled over the >> next 2 days before the subscriber died a horrible death and had to be >> rebuilt. >> >> I believe that /var and /etc/selinux had most of the runaway logs. TAC >> couldn't even create a root support account because the partition was full. >> >> Just a caution to those who patch VMware regularly. Make sure you have >> alerting on filesystem capacity. Ours were 2500 user nodes so there's only >> ~3 GB free on the active partition normally. >> >> I would be very interested if anyone has further info on this bug. >> >> >> >> >> NOTICE TO RECIPIENT: The information contained in this message from >> Great River Energy and any attachments are confidential and intended >> only for the named recipient(s). If you have received this message in >> error, you are prohibited from copying, distributing or using the >> information. Please contact the sender immediately by return email and >> delete the original message. >> >> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> > > > > -- > Ed Leatherman > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VMTools - CUCM 10.5
We just patched vsphere this past weekend and ran into the selinux bug with vmtools, and have also noted some weird stuff with memory and disk space but on Unity Connection. I'll check in with the guy that was handling it and see if there was any other strangeness, this doesn't give me warm and fuzzies. On Wed, Feb 24, 2016 at 9:13 PM, Hughes, Scott GRE-MG wrote: > Has anyone dealt with bugID > CSCul78735 lately? > We just installed our latest round of VMware 5.5 patches. Along with it > came new VMtools. Usual (automatic) upgrade method did not work. The VMware > tools status went from 'out of date' to 'not installed' > > We were able to get the tools installed on our two subscribers by > switching SELinux to permissive mode as the bug describes. We switched them > back to enforcing. Then, we found that the active partition filled over the > next 2 days before the subscriber died a horrible death and had to be > rebuilt. > > I believe that /var and /etc/selinux had most of the runaway logs. TAC > couldn't even create a root support account because the partition was full. > > Just a caution to those who patch VMware regularly. Make sure you have > alerting on filesystem capacity. Ours were 2500 user nodes so there's only > ~3 GB free on the active partition normally. > > I would be very interested if anyone has further info on this bug. > > > > > NOTICE TO RECIPIENT: The information contained in this message from > Great River Energy and any attachments are confidential and intended > only for the named recipient(s). If you have received this message in > error, you are prohibited from copying, distributing or using the > information. Please contact the sender immediately by return email and > delete the original message. > > > > _______ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MOH with CUBE
Thanks Anthony, +1 for the appropriate youtube. On Tue, Feb 9, 2016 at 8:40 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > It's funny you should mention packet counters on your IOS XE platform, as > I was just watching Paul Giralt's talk on SIP troubleshooting (worth the > entire watch, multiple times over). Check out the video at precisely the > 1:00:44 mark, linked below. Your welcome. > <https://www.youtube.com/watch?v=oYZD1sQBdlE> ;) > > BRKUCC-2932 - Troubleshooting SIP with Cisco Unified Communications (2015 > San Diego) - 2 Hours > <https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=83773&backBtn=true> > > On Tue, Feb 9, 2016 at 11:28 AM, Ed Leatherman > wrote: > >> When I first read it I thought he just meant he hated it that much :) >> >> Awesome write-up though Anthony. Now that I see what that midcall >> signalling setting is actually doing, i'm a fan. >> >> re: Troubleshooting this stuff.. I'm trying to use "show call active >> voice [brief]" to see calls in service; none of the packet counters seem to >> work though... dur 00:01:16 tx:0/0 rx:0/0. That seemed to be the command to >> use based on docs I was looking at but i'm not getting any moving counters. >> Is there a different command i'm not finding that will show me these? This >> one is listing each call leg which I kinda like seeing. >> >> On Tue, Feb 9, 2016 at 12:19 PM, Ryan Huff wrote: >> >>> How does one pee on an onion? (clearly, I know you meant PEELING >>> and I am in no way grammar police). >>> >>> I just got a sincere and much needed chuckle from this, "*In my >>> experience, turning up SIP services is like peeing an onion ...*". >>> >>> All in good fun Anthony ;) >>> >>> Sent from my iPhone >>> >>> On Feb 9, 2016, at 12:12 PM, Anthony Holloway < >>> avholloway+cisco-v...@gmail.com> wrote: >>> >>> In my experience, turning up SIP services is like peeing an onion >>> >>> >> >> >> -- >> Ed Leatherman >> > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MOH with CUBE
When I first read it I thought he just meant he hated it that much :) Awesome write-up though Anthony. Now that I see what that midcall signalling setting is actually doing, i'm a fan. re: Troubleshooting this stuff.. I'm trying to use "show call active voice [brief]" to see calls in service; none of the packet counters seem to work though... dur 00:01:16 tx:0/0 rx:0/0. That seemed to be the command to use based on docs I was looking at but i'm not getting any moving counters. Is there a different command i'm not finding that will show me these? This one is listing each call leg which I kinda like seeing. On Tue, Feb 9, 2016 at 12:19 PM, Ryan Huff wrote: > How does one pee on an onion? (clearly, I know you meant PEELING and > I am in no way grammar police). > > I just got a sincere and much needed chuckle from this, "*In my > experience, turning up SIP services is like peeing an onion ...*". > > All in good fun Anthony ;) > > Sent from my iPhone > > On Feb 9, 2016, at 12:12 PM, Anthony Holloway < > avholloway+cisco-v...@gmail.com> wrote: > > In my experience, turning up SIP services is like peeing an onion > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Slightly OT but interesting
Saw this and thought it was pretty cool, figured others on this list might also. Voice app designed to "talk" to telemarketers/robocallers. http://jollyrogertelephone.com/ I've seen other services that just hangs up on them. This one actually sits there and tries to talk to them and connect to a live agent. -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MOH with CUBE
Anthony, Removing the sdp passthru did the trick; now I need to take some debugs and compare so maybe i can understand what was passing thru causing the problem. I'll try that other command you suggested too. I have an exact duplicate of this trunk to turn up in a few weeks so I'm happy to learn this stuff now. Erick - I didn't try the duplex media streaming - wanted to see if there was something I could do locally on CUBE instead of making a cluster-wide change like that... but I was close to trying it. Thanks!! Ed On Mon, Feb 8, 2016 at 5:49 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > I just wanted to comment on two things: > > 1) The port 4000 thing. CUCM does this to just give a port number, it > doesn't actually use it. I wouldn't be looking to hard at that as a > problem. > > *4000 - 4005 / TCP* > *These ports are used as phantom Real-Time Transport Protocol (RTP) and > Real-Time Transport Control Protocol (RTCP) ports for audio, video and data > channel when Cisco Unified Communications Manager does not have ports for > these media.* > *Source: TCP and UDP Port Usage Guide for Cisco Unified Communications > Manager, Release 10.0(1) > <http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/10_0_1/CUCM_BK_T537717B_00_tcp-port-usage-guide-100.html>* > > > With the way SDP works, if the offered port is 4000, and the media > attribute a=sednonly is present, then the port is essentially ignored. > Hence, a half duplex stream, and not full duplex. > > 2) Why are you using this command "pass-thru content sdp" As far as I am > aware, that command will pass thru SDP from CUCM directly to the ITSP. Is > that something you need? Typically, CUBE is your demarc between your > enterprise network and the service provider, and as such, you don't pass > through anything directly. If you don't know why that's there, then I > would recommend removing it and re-testing your MOH scenario. > > A similar command you might want to run is to suppress all of the chatter > CUCM will send to CUBE that really has no business going out to the ITSP, > but keeping the important messages, such as mid-call media changes. > > voice service voip > sip > midcall-signaling passthru media-change > ! > > As far as your 4K router and the closeness of the AS1K defect, I really > don't know. > > On Mon, Feb 8, 2016 at 3:47 PM, Ed Leatherman > wrote: > >> I'm working on getting a SIP trunk with an ITSP fully functional. I can >> get basic calls ok but Unicast MOH is not working out - no audio. Going >> off-hold i get the call audio back. >> >> Quick packet cap on the CUBE confirms i'm getting MOH packets from CUCM >> but they don't make it across CUBE out to the SP. >> >> For the re-INVITE to get the music audio, CUCM is sending SDP with: >> m=audio 4000 RTP/AVP 0 >> >> From the packet cap, the audio packets are not being sourced from port >> 4000 - they are coming in from ephemeral ports. Could this be causing an >> issue with CUBE not translating the streams? >> >> The reason I ask is that I noticed a bug out there CSCtb32219 >> <https://www.cisco.com/cisco/psn/bssprt/bss?searchType=bstbugidsearch&page=bstBugDetail&BugID=CSCtb32219> >> for >> ASR1K which seems close, in my case this is a 4431 (Also ios-xe) running >> 15.5(1)S. Anyone run into that? The workaround is to enable duplex >> streaming in CUCM, which seems a little goofy. >> >> I dont feel like I have anything special configured on CUBE: >> voice service voip >> ip address trusted list >> ipv4 blahblahblah >> address-hiding >> allow-connections sip to sip >> no supplementary-service sip refer >> fax protocol pass-through g711ulaw >> sip >> pass-thru content sdp >> sip-profiles 100 >> ! >> >> dialpeers all have >> ! >> >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> >> >> Thanks! >> >> >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MOH with CUBE
Hi Anthony, For #2.. Its just there because the sp's integration guide had it in there- I'll try it tomorrow without and see if it fixes it, what you say makes sense. thanks! On Mon, Feb 8, 2016, 5:49 PM Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > I just wanted to comment on two things: > > 1) The port 4000 thing. CUCM does this to just give a port number, it > doesn't actually use it. I wouldn't be looking to hard at that as a > problem. > > *4000 - 4005 / TCP* > *These ports are used as phantom Real-Time Transport Protocol (RTP) and > Real-Time Transport Control Protocol (RTCP) ports for audio, video and data > channel when Cisco Unified Communications Manager does not have ports for > these media.* > *Source: TCP and UDP Port Usage Guide for Cisco Unified Communications > Manager, Release 10.0(1) > <http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/10_0_1/CUCM_BK_T537717B_00_tcp-port-usage-guide-100.html>* > > > With the way SDP works, if the offered port is 4000, and the media > attribute a=sednonly is present, then the port is essentially ignored. > Hence, a half duplex stream, and not full duplex. > > 2) Why are you using this command "pass-thru content sdp" As far as I am > aware, that command will pass thru SDP from CUCM directly to the ITSP. Is > that something you need? Typically, CUBE is your demarc between your > enterprise network and the service provider, and as such, you don't pass > through anything directly. If you don't know why that's there, then I > would recommend removing it and re-testing your MOH scenario. > > A similar command you might want to run is to suppress all of the chatter > CUCM will send to CUBE that really has no business going out to the ITSP, > but keeping the important messages, such as mid-call media changes. > > voice service voip > sip > midcall-signaling passthru media-change > ! > > As far as your 4K router and the closeness of the AS1K defect, I really > don't know. > > On Mon, Feb 8, 2016 at 3:47 PM, Ed Leatherman > wrote: > >> I'm working on getting a SIP trunk with an ITSP fully functional. I can >> get basic calls ok but Unicast MOH is not working out - no audio. Going >> off-hold i get the call audio back. >> >> Quick packet cap on the CUBE confirms i'm getting MOH packets from CUCM >> but they don't make it across CUBE out to the SP. >> >> For the re-INVITE to get the music audio, CUCM is sending SDP with: >> m=audio 4000 RTP/AVP 0 >> >> From the packet cap, the audio packets are not being sourced from port >> 4000 - they are coming in from ephemeral ports. Could this be causing an >> issue with CUBE not translating the streams? >> >> The reason I ask is that I noticed a bug out there CSCtb32219 >> <https://www.cisco.com/cisco/psn/bssprt/bss?searchType=bstbugidsearch&page=bstBugDetail&BugID=CSCtb32219> >> for >> ASR1K which seems close, in my case this is a 4431 (Also ios-xe) running >> 15.5(1)S. Anyone run into that? The workaround is to enable duplex >> streaming in CUCM, which seems a little goofy. >> >> I dont feel like I have anything special configured on CUBE: >> voice service voip >> ip address trusted list >> ipv4 blahblahblah >> address-hiding >> allow-connections sip to sip >> no supplementary-service sip refer >> fax protocol pass-through g711ulaw >> sip >> pass-thru content sdp >> sip-profiles 100 >> ! >> >> dialpeers all have >> ! >> >> dtmf-relay rtp-nte >> codec g711ulaw >> no vad >> >> >> Thanks! >> >> >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] MOH with CUBE
I'm working on getting a SIP trunk with an ITSP fully functional. I can get basic calls ok but Unicast MOH is not working out - no audio. Going off-hold i get the call audio back. Quick packet cap on the CUBE confirms i'm getting MOH packets from CUCM but they don't make it across CUBE out to the SP. For the re-INVITE to get the music audio, CUCM is sending SDP with: m=audio 4000 RTP/AVP 0 >From the packet cap, the audio packets are not being sourced from port 4000 - they are coming in from ephemeral ports. Could this be causing an issue with CUBE not translating the streams? The reason I ask is that I noticed a bug out there CSCtb32219 <https://www.cisco.com/cisco/psn/bssprt/bss?searchType=bstbugidsearch&page=bstBugDetail&BugID=CSCtb32219> for ASR1K which seems close, in my case this is a 4431 (Also ios-xe) running 15.5(1)S. Anyone run into that? The workaround is to enable duplex streaming in CUCM, which seems a little goofy. I dont feel like I have anything special configured on CUBE: voice service voip ip address trusted list ipv4 blahblahblah address-hiding allow-connections sip to sip no supplementary-service sip refer fax protocol pass-through g711ulaw sip pass-thru content sdp sip-profiles 100 ! dialpeers all have ! dtmf-relay rtp-nte codec g711ulaw no vad Thanks! -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] CUCM 10.x Application Dial Rules - When do they get used?
The documents i've seen don't give a definitive list of what cases these get applied - it just says "applications such as". I want to add in some rules here for Jabber click to call stuff, for example changing 1+10 digits to sometime dialable in our current plan. I'm being a little cautious about unintended consequences. IPMA Cisco Web Dialer Jabber - Click to call, what other cases? Remote Destinations Is there a rule that gets checked to see when these rules get applied? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX/Extend and Connect
Figured out my issue - for posterity: Agent DN on the CTI RD was in a partition specifically for UCCX Agents, which cannot typically be dialed by most CSS's. The CTI RD device itself was not configured to be able to call its own DN. I usually don't allow agents to call each other's agent lines since I'm under the impression this is one of the myriad of unsupported configurations. This resulted in a CTI timeout in the UCCX JTAPI logs, tracked back to SDL trace on the CUCM node one which the CTI Remote Device was registered. Digit analysis failed against the agent DN, where the CSS's that CM was using were the concat of the device/line CSS of the CTI RD DN. So CM is using the CTI remote device configuration of the device to dial it's own DN. Headache! Changing the CSS configuration of the CTI RD Device/DN to allow it to call Agent DNs allows it to work as advertised. IM&P wasn't necessary. On Tue, Jan 26, 2016 at 7:04 PM, Brian V wrote: > In my experience you don't need IM&P to make extend and connect work. > > On Tue, Jan 26, 2016 at 2:44 PM, Ed Leatherman > wrote: > >> Trying to get this feature working - is IM&P actually required here for >> something? We're just phone-only jabber right now, no IMP server. >> >> I have the CTI RD setup, associated with RMCM user, I can log into >> jabber, set remote number, sign into Finesse. When I go ready, it does not >> extend the persistent call. When a call is presented, it tries to hit my >> remote number but doesn't actually connect. UCCX is send to enable the >> persistent connection. >> >> Only difference I can tell is I don't have IMP running - seems odd if >> that is required, what does it use that for? >> >> UCCX 10.6, CUCM 10.5 >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX/Extend and Connect
The docs on the UCCX side of the house for this were kinda bad imo - they just kinda threw out some links to CUCM and then gave a 30,000 foot view procedure of installing IM&P which sort of set off my WTF alarm so I decided to just try it w/o IMP; vanilla extend and connect works just dandy without it. On Tue, Jan 26, 2016 at 11:51 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > I'm replying tongue in cheek, as I've never done it before, so I have no > idea. But, as someone who would go straight to the documentation on this, > I found this to be comical, in light of your reply. > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] UCCX/Extend and Connect
Trying to get this feature working - is IM&P actually required here for something? We're just phone-only jabber right now, no IMP server. I have the CTI RD setup, associated with RMCM user, I can log into jabber, set remote number, sign into Finesse. When I go ready, it does not extend the persistent call. When a call is presented, it tries to hit my remote number but doesn't actually connect. UCCX is send to enable the persistent connection. Only difference I can tell is I don't have IMP running - seems odd if that is required, what does it use that for? UCCX 10.6, CUCM 10.5 -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] RTP ports used by phones
I did not catch the port that it was receiving on, must have been a valid port as it didn't bounce anywhere that I could find in syslog. On Tue, Jan 19, 2016 at 4:10 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > The Phone would offer up it's own UDP port of...in your case...32768, to > CUCM, then CUCM converts this to an MGCP message and sends to the gateway. > This way the gateway knows where to send RTP. So, if anything, it's the > phone's fault for offering up 32768, not the gateway. Did you happen to > catch what port the gateway was receiving RTP on? > > On Tue, Jan 19, 2016 at 12:55 PM, Ed Leatherman > wrote: > >> Come to think of it, the session was between an IOS MGCP gateway and the >> 8945 - perhaps something the gateway is trying to send to the phone that I >> dont realize. >> >> On Tue, Jan 19, 2016 at 11:54 AM, Ryan Huff wrote: >> >>> Ed, >>> >>> >>> >>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/10_0_1/CUCM_BK_T537717B_00_tcp-port-usage-guide-100/CUCM_BK_T537717B_00_tcp-port-usage-guide-100_chapter_01.html >>> >>> >>> Could it be that you're seeing something from the ephemeral port range >>> 32768 – 61000? >>> >>> >>> Thanks, >>> >>> >>> = Ryan = >>> >>> >>> >>> Email: ryanthomash...@outlook.com >>> >>> Spark: ryanthomash...@outlook.com >>> >>> Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> >>> >>> LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff> >>> >>> Web ryanthomashuff.com >>> >>> >>> -- >>> *From:* cisco-voip on behalf of Ed >>> Leatherman >>> *Sent:* Tuesday, January 19, 2016 11:46 AM >>> *To:* Cisco VOIP >>> *Subject:* [cisco-voip] RTP ports used by phones >>> >>> I've notice this a few times bouncing on ACL, thought it was worth >>> asking about. >>> >>> I see in numerous documentation that CUCM uses 16384 - 32767 for RTP - >>> the documents specifically say IP Phone to IPVMS. >>> >>> I observed an 8945 Cisco phone listening on 32768 and 32769 (assuming >>> RTP and associated RTCP) due to access list not permitting it. Is there a >>> doc somewhere that shows different/expanded range of ports that Cisco >>> phones will use? >>> >>> >>> >>> -- >>> Ed Leatherman >>> >> >> >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] RTP ports used by phones
Thanks Anthony, I should have looked in the SRND - I was googling around for port utilization. Your explanation sounds plausible, it is a SCCP phone and I've never seen any other port numbers pop up like that. In this case the ACL is actually just running on a 4500X, no real firewall in play to do inspection.. so i'll probably just revise and get it queued up to push out to all the devices that run that acl. Thanks! On Tue, Jan 19, 2016 at 4:08 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Is this a SCCP or SIP phone? I wouldn't worry too much about where you > saw "IPVMS", it's likely specific to the context of the chapter/section. > I.g., Table 6 in Port Usage Guide is only for Phone to CUCM communication > > *For SCCP Phones, From the SRND* > > *SCCP endpoints use a non-configurable hard-coded range of 16384 to 32767 > for voice-only calls* > > *For SIP Phones, From the CUCM Administration Guide* > > *SIP Profile* > *Start Media Port = This field designates the start real-time protocol > (RTP) port for media. Media port ranges from 2048 to 65535. Default > specifies 16384.* > *Stop Media Port = This field designates the stop real-time protocol (RTP) > port for media. Media port ranges from 2048 to 65535. Default specifies > 32766. * > > *For Gateways, From Port Usage Guide* > > *Gateway to Unified Communications Manager 16384 - 32767 / UDP* > > First recommendation, is to use deep packet inspection and let the UDP > ports be opened by the firewall dynamically. This works with MGCP, H323, > SIP and SCCP. > > > http://www.cisco.com/c/en/us/td/docs/security/asa/asa82/configuration/guide/config/inspect_voicevideo.html > > Second recommendation, is to use SIP signaling so you can control the RTP > port range so that it always matches your configured ACLs exactly. > > And in closing, I think there is probably some confusion around what the > exact upper bound is: 32766 or 32768, and you probably found a phone model > firmware that thought 32768 was the upper bound. I personally, have always > gone with 32766 as being the upper bound, but then again, I've never > created an ACL for this range either, so it hasn't presented itself as a > problem thus far. > > On Tue, Jan 19, 2016 at 10:46 AM, Ed Leatherman > wrote: > >> I've notice this a few times bouncing on ACL, thought it was worth asking >> about. >> >> I see in numerous documentation that CUCM uses 16384 - 32767 for RTP - >> the documents specifically say IP Phone to IPVMS. >> >> I observed an 8945 Cisco phone listening on 32768 and 32769 (assuming RTP >> and associated RTCP) due to access list not permitting it. Is there a doc >> somewhere that shows different/expanded range of ports that Cisco phones >> will use? >> >> >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] RTP ports used by phones
Come to think of it, the session was between an IOS MGCP gateway and the 8945 - perhaps something the gateway is trying to send to the phone that I dont realize. On Tue, Jan 19, 2016 at 11:54 AM, Ryan Huff wrote: > Ed, > > > > http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/10_0_1/CUCM_BK_T537717B_00_tcp-port-usage-guide-100/CUCM_BK_T537717B_00_tcp-port-usage-guide-100_chapter_01.html > > > Could it be that you're seeing something from the ephemeral port range > 32768 – 61000? > > > Thanks, > > > = Ryan = > > > > Email: ryanthomash...@outlook.com > > Spark: ryanthomash...@outlook.com > > Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> > > LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff> > > Web ryanthomashuff.com > > > -- > *From:* cisco-voip on behalf of Ed > Leatherman > *Sent:* Tuesday, January 19, 2016 11:46 AM > *To:* Cisco VOIP > *Subject:* [cisco-voip] RTP ports used by phones > > I've notice this a few times bouncing on ACL, thought it was worth asking > about. > > I see in numerous documentation that CUCM uses 16384 - 32767 for RTP - the > documents specifically say IP Phone to IPVMS. > > I observed an 8945 Cisco phone listening on 32768 and 32769 (assuming RTP > and associated RTCP) due to access list not permitting it. Is there a doc > somewhere that shows different/expanded range of ports that Cisco phones > will use? > > > > -- > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] RTP ports used by phones
I've notice this a few times bouncing on ACL, thought it was worth asking about. I see in numerous documentation that CUCM uses 16384 - 32767 for RTP - the documents specifically say IP Phone to IPVMS. I observed an 8945 Cisco phone listening on 32768 and 32769 (assuming RTP and associated RTCP) due to access list not permitting it. Is there a doc somewhere that shows different/expanded range of ports that Cisco phones will use? -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 11 and Finesse
Also related.. TAC advised me that CSCuv24965 also is present in v11 - the bug report doesn't list 11 yet, i just logged a new ticket to ask why. Basically it will randomly wipe out all your reason codes and any customization you have in finesse for desktops - in my case it was all my workarounds from the that technote Anthony linked. If you hit the bug, be prepared to restart Finesse Tomcat and re-enter in all your finesse customization. On Wed, Jan 13, 2016 at 6:42 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Just a heads up to my Cisco Collab brothers and sisters. > > If you are deploying UCCX v11 with Enhanced or Standard licensing, you can > expect Live Data to be broken in Finesse. > > Cisco TAC has to root your box and upload new JAR files to fix it. > > https://tools.cisco.com/bugsearch/bug/cscux33949 > > And also, not a defect, but a feature which would be undesirable by most: > Finesse Supervisors with Reporting Role see all CSQs instead of just their > Team only (not specific to v11). > > > http://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express/118823-technote-uccx-00.html > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX Bug
Tac engineer said it was an ongoing issue that they thought was fixed in one of the 10.6su's but turns out not, doesn't appear that they know exactly what root cause is but sounds like maybe DRS related. I'm out of the office at the moment but I believe we are on latest su+es something. I forgot about the custom XML ugh, thx for pointing that out. We only made minor changes, luckily I copied it onto our internal wiki. On Tue, Dec 29, 2015, 12:22 PM Justin Steinberg wrote: > are you running 10.6(1)su1 ?I have seen this on 10.6 base on two > seperate installs. it's really ugly, all Finesse Admin team specific data > is lost. the biggest problem is if you customize XML and then lose it and > have to try and figure out how to rebuild it. Since i had this the first > time I make it a point to save the XML desktop layout on my PC for each > team in the event I have to restore. > > The first time I had this problem was probably close to a year ago i'd > guess, but i personally haven't seen it on 10.(6)1su1. > > On Tue, Dec 29, 2015 at 9:45 AM, Ed Leatherman > wrote: > >> Just hit this fun bug, thought I would pass it along. CSCuv24965 >> >> Running latest uccx 10.6, HA, enhanced. >> >> TAC said it currently had a lot of attention with the developers, no fix >> yet and no idea yet what triggers it. Workaround is restarting tomcat and >> re-associating all of your reason codes and other team related settings >> with the correct teams. >> >> Happy Holidays! :) >> >> -- >> Ed Leatherman >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] UCCX Bug
Just hit this fun bug, thought I would pass it along. CSCuv24965 Running latest uccx 10.6, HA, enhanced. TAC said it currently had a lot of attention with the developers, no fix yet and no idea yet what triggers it. Workaround is restarting tomcat and re-associating all of your reason codes and other team related settings with the correct teams. Happy Holidays! :) -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] VCS Checker
Thats really neat good find On Wed, Dec 9, 2015 at 4:13 PM, Ryan Huff wrote: > Not sure when Cisco made this tool public but it is a welcomed edition! I > knew TAC had it but didn't realize Cisco opened it up. > > > https://cway.cisco.com/tools/SrvRecord > > > = Ryan = > > > > Email: ryanthomash...@outlook.com > > Spark: ryanthomash...@outlook.com > > Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> > > LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff> > > Web ryanthomashuff.com > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX + CUIC 10.6: Understanding Permissions
So what i had to do today (UCCX 10): CUIC user created a dashboard that she wanted to share with her team. BUT users can't do that themselves (AFAIK). I logged in as the uccxadmin account, created a user group for that team, added the required users to it, assigned execute and write permissions to the dashboard and all the reports it used (individually) to the user group i created. WHat I did as far as subcategories - from memory - was log in as the uccxadmin account, create a "My Reports" subcategory off of reports and a similar one off of dashboards, and futz with the permissions so that anyone would write to them. Then the cuic users could create reports/dashs in that folder - they still can only see/run reports they created though. To share the same report I had to do as above as admin user. Curious if there's better way to do these things though. On Tue, Dec 8, 2015 at 11:44 AM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > All, > > I have to admit defeat on CUIC permissions. I just don't get it yet. I > thought I understood it, but things often end up greyed out for myself, and > it frustrates me that I cannot grasp this concept. > > What is a good explanation for how permissions work? What are some basic > permission settings I can look at/use to just get some basic functionality > working? What is the best document to reference? > > I'm not even to the hierarchical design of permissions yet, I'm still just > trying to figure out a good flat structure to start. E.g., New users not > being able to create subcategories out of the gate. > > Am I alone in thinking this: CUIC a bit more complicated than it needs to > be? > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Informacast
With the paging stuff, i've found if I'm stuck its usually either i've screwed up mcast or I've forgotten to configure the backend user account/device associations in call manager. If I dial the paging number and the phones beep/open speakers and then no audio, its the former. If I dial the paging number and nothing happens at all, it's likely at least the latter (and maybe also the former :) ) On Mon, Dec 7, 2015 at 2:26 PM, Aaron Banks wrote: > I've never configured Informacast before. I have most of the setup > complete, the only piece I'm hung up on is the 3850 switch (which is > stacked). I want to enable multicast but it is only for the voice vlan. > There is no routing being done between vlans or multiple switches. Do I > use ip multicast routing to enable multicasting globally or ip multicast > auto-enable? Also, is PIM or IGMP normally used? IGMP is already running, > so I was thinking I only had to configure the multicast address for it. > > Aaron > > > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Did Cisco Remove all voice downloads from Cisco.com?
Too late, Lelio already suited up and took off... he was hoping to find some forwarding on non-primary lines for his 7960's while he's out there. >:) On Mon, Dec 7, 2015 at 11:50 AM, Ryan Ratliff (rratliff) wrote: > Looks like anything under the “Unified Communications” umbrella could have > been affected. You can call off the black helicopters :) > > -Ryan > > On Dec 7, 2015, at 10:21 AM, Ryan Huff wrote: > > It makes more sense that was a snafu or some sort of a planned > maintenance. It is odd though, that it seemed to only selectively impact > some of the collaboration images, specifically those that are going the > cloud, and that it happened the weekend before the summit. I dont recall > seeing a maintenance notice. > > ---- Original message > From: Ed Leatherman > Date:12/07/2015 9:52 AM (GMT-05:00) > To: Ryan Huff > Cc: Charles Goldsmith ,cisco-voip@puck.nether.net > Subject: Re: [cisco-voip] Did Cisco Remove all voice downloads from > Cisco.com <http://cisco.com>? > > This sounds ominous ! > > On Sun, Dec 6, 2015 at 7:56 PM, Ryan Huff wrote: > >> Hold your ISO's tight >> >> I am withholding judgment and opinion till after the collaboration summit >> and Rowan & Jonathan have a chance to lay the plan out. Some of the pre >> released stuff looks interesting though ... >> >> I would theorize, and especially in the case of an active sntc, a call to >> TAC would get you what you need, for the time being anyway. I haven't >> tested that theory mind you, but I just don't see Cisco turning off the >> firehose mid-stream without warning and saying, "too bad so sad". >> >> -Ryan >> >> ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Did Cisco Remove all voice downloads from Cisco.com?
This sounds ominous ! On Sun, Dec 6, 2015 at 7:56 PM, Ryan Huff wrote: > Hold your ISO's tight > > I am withholding judgment and opinion till after the collaboration summit > and Rowan & Jonathan have a chance to lay the plan out. Some of the pre > released stuff looks interesting though ... > > I would theorize, and especially in the case of an active sntc, a call to > TAC would get you what you need, for the time being anyway. I haven't > tested that theory mind you, but I just don't see Cisco turning off the > firehose mid-stream without warning and saying, "too bad so sad". > > -Ryan > > ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Expressway ?'s
Followup - Not possible how I'm trying to get around this; After the client goes into edge mode, it sends a request to expressway: get_edge_config?service_name=_cisco-uds&service_name=_cuplogin Then expressway does the SRV lookup for cisco-uds. So that SRV still must exist somewhere for expressway to get it and learn the right cucm etc to send the client to (i'm assuming at this point). So i still need to solve my split DNS issue :) On Fri, Nov 6, 2015 at 8:32 AM, Ed Leatherman wrote: > This thread is so much fun I thought i'd resurrect it again! > > I'm toying around with MRA again, as we have a significant segment of our > campus that has their hosts all NATed off from the rest of campus/world and > their firewalls aren't playing well with SIP. > > As I don't have a ready mechanism to do split DNS, I thought perhaps I > could just make a new service domain (jabber.wvu.edu) and only create the > collab-edge SRV for that. Idea would be if I signed into jabber as > ealeather...@jabber.wvu.edu it would force it over to expressway. Running > into lots of challenges here and starting to wonder if this is even > possible. > > Right now exp doesn't seem to be able to locate UDS for me. I'm getting > some certificate errors in expC for my call managers: > > edgeconfigprovisioning: Level="ERROR > <https://expc.telecom.wvu.edu/eventlog?all_text=TGV2ZWw9IkVSUk9SIg==>" > Detail="Certificate verify failure > <https://expc.telecom.wvu.edu/eventlog?all_text=RGV0YWlsPSJDZXJ0aWZpY2F0ZSB2ZXJpZnkgZmFpbHVyZSI=>" > Server="< > <https://expc.telecom.wvu.edu/eventlog?all_text=U2VydmVyPSIxMC4xOTIuMi4xMiI=>server > IP>" Reason="No subject alternate name > <https://expc.telecom.wvu.edu/eventlog?all_text=UmVhc29uPSJObyBzdWJqZWN0IGFsdGVybmF0ZSBuYW1lIg==>" > UTCTime="2015-11-06 13:25:29,611 > <https://expc.telecom.wvu.edu/eventlog?all_text=VVRDVGltZT0iMjAxNS0xMS0wNiAxMzoyNToyOSw2MTEi> > " > > Is it acting like this because my server names in CUCM are defined as IP > Addresses and not host names? this seems strange though, on the Unified CM > servers page in expc all my nodes are listed and TLS and TCP both say > active. > > anyone have 2 cents to give on this one? > > > > > On Thu, Jun 18, 2015 at 10:01 PM, Lelio Fulgenzi > wrote: > >> I really wish there was another option other than split DNS to get MRA >> working from off-premise. I mean, why rely on DNS response rather than lack >> of connectivity to decide which path to take? A parameter in the >> jabber-config.xml file could help with that. >> >> Anyways, I know it's gonna be fun to use the workaround of configuring >> our edge firewall to filter out DNS responses. ugh. >> >> --- >> Lelio Fulgenzi, B.A. >> Senior Analyst, Network Infrastructure >> Computing and Communications Services (CCS) >> University of Guelph >> >> 519‐824‐4120 Ext 56354 >> le...@uoguelph.ca >> www.uoguelph.ca/ccs >> Room 037, Animal Science and Nutrition Building >> Guelph, Ontario, N1G 2W1 >> >> -- >> *From: *"Charles Goldsmith" >> *To: *"Scott Voll" >> *Cc: *cisco-voip@puck.nether.net >> *Sent: *Thursday, 18 June, 2015 7:45:14 PM >> *Subject: *Re: [cisco-voip] Expressway ?'s >> >> >> As said by others, license is free for the MRA part, to get the free >> license, here is a handy blog entry : >> https://ciscocollab.wordpress.com/2014/02/20/how-to-get-expressway-c-and-e-licenses/ >> >> >> He also has entries on helping set it up, but it's pretty simple once you >> get in and start configuring. Hard part is getting the certs, DNS and >> firewall in line :) >> >> >> On Thu, Jun 18, 2015 at 4:58 PM, Scott Voll wrote: >> >>> I"m still on UC 8.6. we are planning an upgrade to 10.x We currently >>> have DLU's for licensing and will be moving to CUWL Standard ( I think). >>> >>> How does Expressways factor into this? >>> >>> is it part of CUWL? Is there a Cost? What all can you do with >>> Expressway. What I believe I understand is that it can get your external >>> voice and video internal. does it replace my lan to lan connections to get >>> an IP phone registered to CM? >>> >>> Does it also do video bridging? Example. Polycom HDX unit, cisco SX20, >>> jabber and skype all in a single call? >>> >>> TIA >>> >>> Scott >>> >>> >>> >>> ___ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > > -- > Ed Leatherman > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Speech Connect, Unity Con 10.5
Good morning! Anyone know if it is possible somehow to modify the subsequent prompts (not the initial greeting) in the speech connect directory handler? I see on Ciscounitytools.com that there are some alternate greetings available for this, and where it can be changed in cuadmin.. however I have an internal customer interested in changing the subsequent prompts for no matches. There seem to be two of them and they are inconsistent. Also they refer to dialing 0 for operator and we do not have an operator to send them to, so the concern with the customer is that the caller will think they are going to get a live handoff when instead i'll have to send them to another IVR application somewhere instead. Doesn't appear to be possible but figured I'd ask in case there's a nerd knob somewhere that I missed. -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] CUCM losing access to NTP for extended period
Forwarding this to the list for Lelio: what happens if our voice servers loose access to ntp servers for a long period of time? His Internet service is down -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Expressway ?'s
This thread is so much fun I thought i'd resurrect it again! I'm toying around with MRA again, as we have a significant segment of our campus that has their hosts all NATed off from the rest of campus/world and their firewalls aren't playing well with SIP. As I don't have a ready mechanism to do split DNS, I thought perhaps I could just make a new service domain (jabber.wvu.edu) and only create the collab-edge SRV for that. Idea would be if I signed into jabber as ealeather...@jabber.wvu.edu it would force it over to expressway. Running into lots of challenges here and starting to wonder if this is even possible. Right now exp doesn't seem to be able to locate UDS for me. I'm getting some certificate errors in expC for my call managers: edgeconfigprovisioning: Level="ERROR <https://expc.telecom.wvu.edu/eventlog?all_text=TGV2ZWw9IkVSUk9SIg==>" Detail="Certificate verify failure <https://expc.telecom.wvu.edu/eventlog?all_text=RGV0YWlsPSJDZXJ0aWZpY2F0ZSB2ZXJpZnkgZmFpbHVyZSI=>" Server="< <https://expc.telecom.wvu.edu/eventlog?all_text=U2VydmVyPSIxMC4xOTIuMi4xMiI=>server IP>" Reason="No subject alternate name <https://expc.telecom.wvu.edu/eventlog?all_text=UmVhc29uPSJObyBzdWJqZWN0IGFsdGVybmF0ZSBuYW1lIg==>" UTCTime="2015-11-06 13:25:29,611 <https://expc.telecom.wvu.edu/eventlog?all_text=VVRDVGltZT0iMjAxNS0xMS0wNiAxMzoyNToyOSw2MTEi> " Is it acting like this because my server names in CUCM are defined as IP Addresses and not host names? this seems strange though, on the Unified CM servers page in expc all my nodes are listed and TLS and TCP both say active. anyone have 2 cents to give on this one? On Thu, Jun 18, 2015 at 10:01 PM, Lelio Fulgenzi wrote: > I really wish there was another option other than split DNS to get MRA > working from off-premise. I mean, why rely on DNS response rather than lack > of connectivity to decide which path to take? A parameter in the > jabber-config.xml file could help with that. > > Anyways, I know it's gonna be fun to use the workaround of configuring our > edge firewall to filter out DNS responses. ugh. > > --- > Lelio Fulgenzi, B.A. > Senior Analyst, Network Infrastructure > Computing and Communications Services (CCS) > University of Guelph > > 519‐824‐4120 Ext 56354 > le...@uoguelph.ca > www.uoguelph.ca/ccs > Room 037, Animal Science and Nutrition Building > Guelph, Ontario, N1G 2W1 > > -- > *From: *"Charles Goldsmith" > *To: *"Scott Voll" > *Cc: *cisco-voip@puck.nether.net > *Sent: *Thursday, 18 June, 2015 7:45:14 PM > *Subject: *Re: [cisco-voip] Expressway ?'s > > > As said by others, license is free for the MRA part, to get the free > license, here is a handy blog entry : > https://ciscocollab.wordpress.com/2014/02/20/how-to-get-expressway-c-and-e-licenses/ > > > He also has entries on helping set it up, but it's pretty simple once you > get in and start configuring. Hard part is getting the certs, DNS and > firewall in line :) > > > On Thu, Jun 18, 2015 at 4:58 PM, Scott Voll wrote: > >> I"m still on UC 8.6. we are planning an upgrade to 10.x We currently >> have DLU's for licensing and will be moving to CUWL Standard ( I think). >> >> How does Expressways factor into this? >> >> is it part of CUWL? Is there a Cost? What all can you do with >> Expressway. What I believe I understand is that it can get your external >> voice and video internal. does it replace my lan to lan connections to get >> an IP phone registered to CM? >> >> Does it also do video bridging? Example. Polycom HDX unit, cisco SX20, >> jabber and skype all in a single call? >> >> TIA >> >> Scott >> >> >> >> ___ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCx 10.6 CAD vs Finesse
We're running both, you need to load cop file for mixed mode or I think the latest SU has mixed mode built in. It's still only supported team by team iirc - so a team is either all using CAD or all using finesse. On Tue, Oct 27, 2015 at 12:10 PM, Scott Voll wrote: > Does anyone have CAD and Finesse running at the same time? Can you? We > have seen documentation that you can only run one or the other. But I seem > to remember that someone said in 10.6 you can run both. > > Anyone able to confirm or deny? > > TIA > > Scott > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUBE - Dial-peer multiple destination-pattern matching
I don't have any experience with the file based patterns. If they have room for a few more VM's, putting SME in the middle and hang CUBE off of that might be another way to do it cleanly. Then ils could take care of those patterns. On Mon, Oct 26, 2015 at 5:42 AM, Boon wrote: > I have a client who is planning on splitting their single CUCM cluster > with CUBE and PSTN SIP into two separate clusters. > > The challenge is that they want to share the CUBE solution and DID range > between both clusters. > > I can see an opportunity here to use the IOS dial-peer feature 'Multiple > Destination Pattern' matching using a file hosted in the router flash. > > Although the configuration looks pretty straight forward I wanted to find > out if any of you guys had deployed this feature and whether there are any > gotchas to be aware of? I'm aware of the minimum IOS version requirement. > I'm wondering whether file maintenance can become an issue. > > Also, has anyone used this with a CUBE HA solution? > > Any help appreciated. Thanks > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
