[cisco-voip] Extension mobility boot/register to prompt
I have extension mobility setup currently with a fake DN so that the phone will register. If you hit service, you can login and extension mobility works, but is there anyway to make the phone when it boots up come up with the login prompt? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Interested in buying the latest Cisco IP phones
The Cisco 88xx series phones are the most mainstream ones currently. We standardized on the 8861 a few years ago. The Wifi support was a big plus especially with work from home MRA deployment. https://www.cisco.com/c/en/us/products/collaboration-endpoints/ip-phones/index.html#~global-leader -Original Message- From: cisco-voip On Behalf Of Turritopsis Dohrnii Teo En Ming Sent: Friday, August 12, 2022 5:24 AM To: cisco-voip@puck.nether.net Cc: c...@teo-en-ming-corp.com Subject: [cisco-voip] Interested in buying the latest Cisco IP phones Subject: Interested in buying the latest Cisco IP phones Good day from Singapore, I have 4 Cisco 7960 IP phones. I believe almost all of them are faulty because there are black patches on the LCD screens. Perhaps due to the humid environment in Singapore. I am interested in buying new Cisco IP phones. What are the latest models? Thank you. Regards, Mr. Turritopsis Dohrnii Teo En Ming Targeted Individual in Singapore 12 Aug 2022 Fri Blogs: https://tdtemcerts.blogspot.com https://tdtemcerts.wordpress.com ___ 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] [External] Re: expressway E GoDaddy certificate
TAC is the one that showed me. The big clue is that in expressway it has the ability to upload a private key. Why have that feature if you can't extract it? -Original Message- From: Lelio Fulgenzi Sent: Wednesday, August 3, 2022 9:11 AM To: Matthew Huff ; Hunter Fuller Cc: Cisco VOIP Subject: RE: [cisco-voip] [External] Re: expressway E GoDaddy certificate Curious if you passed this method by Cisco/Expressway support. I find the Expressway support team very critical of any changes to supported methods. It's the only team that doesn't support ESXi maintenance releases unless it's explicitly stated in the document. -Original Message- From: Matthew Huff Sent: Wednesday, August 3, 2022 7:47 AM To: Hunter Fuller ; Lelio Fulgenzi Cc: Cisco VOIP Subject: RE: [cisco-voip] [External] Re: expressway E GoDaddy certificate CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca Same. We have a multi-san certificate for our expressway-e cluster from Entrust. You have to create the CSR on the first node in the cluster, install the certificate and then copy the private key via SCP. You then load the private key and certificate into the 2nd server. To get the private key. Login to the server that has the installed certificate via SCP as root. The file is privkey.pem in /tandberg/persistent/certs/ -Original Message- From: cisco-voip On Behalf Of Hunter Fuller Sent: Tuesday, August 2, 2022 1:37 PM To: Lelio Fulgenzi Cc: Cisco VOIP Subject: Re: [cisco-voip] [External] Re: expressway E GoDaddy certificate Since I just love being contrarian, we are running the same cert on both Expressway-E. It is not GoDaddy though. But feel free to take a look at how this works. Our expe are vbhexpe.voip.uah.edu and libexpe.voip.uah.edu and I've also attached the cert to this email. -- Hunter Fuller (they) Router Jockey VBH M-1C +1 256 824 5331 Office of Information Technology The University of Alabama in Huntsville Network Engineering On Tue, Aug 2, 2022 at 9:06 AM Lelio Fulgenzi wrote: > > We’ve always been weary of wildcard and muti-San certs that preclude a > certificate for each server. In our case, we have got a multi-san cert for > each expressway E (and C for that matter) which includes the server as the > primary host, and the peer, cluster name and domain as a SAN. > > > > I’m lucky that our cert team has got a contract with good inventory, so, a > couple of extra multi-SAN certs isn’t a big deal for us. > > > > At some point, we may consider moving the Expressways to Let’s Encrypt. It’s > the only Cisco collab platform that supports it for now. > > > > > > From: cisco-voip On Behalf Of > Shaihan Jaffrey > Sent: Tuesday, August 2, 2022 4:21 AM > To: Cisco VOIP > Subject: [cisco-voip] expressway E GoDaddy certificate > > > > CAUTION: This email originated from outside of the University of > Guelph. Do not click links or open attachments unless you recognize > the sender and know the content is safe. If in doubt, forward > suspicious emails to ith...@uoguelph.ca > > > > what is the process to renew Public certificate on Expressway E > through > > GoDaddy. > > Is one certificate sufficient for primary and secondary exp-e? > > > > do we have to get certificates based on FQDN? > > > > Regards > > ___ > 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] [External] Re: expressway E GoDaddy certificate
Same. We have a multi-san certificate for our expressway-e cluster from Entrust. You have to create the CSR on the first node in the cluster, install the certificate and then copy the private key via SCP. You then load the private key and certificate into the 2nd server. To get the private key. Login to the server that has the installed certificate via SCP as root. The file is privkey.pem in /tandberg/persistent/certs/ -Original Message- From: cisco-voip On Behalf Of Hunter Fuller Sent: Tuesday, August 2, 2022 1:37 PM To: Lelio Fulgenzi Cc: Cisco VOIP Subject: Re: [cisco-voip] [External] Re: expressway E GoDaddy certificate Since I just love being contrarian, we are running the same cert on both Expressway-E. It is not GoDaddy though. But feel free to take a look at how this works. Our expe are vbhexpe.voip.uah.edu and libexpe.voip.uah.edu and I've also attached the cert to this email. -- Hunter Fuller (they) Router Jockey VBH M-1C +1 256 824 5331 Office of Information Technology The University of Alabama in Huntsville Network Engineering On Tue, Aug 2, 2022 at 9:06 AM Lelio Fulgenzi wrote: > > We’ve always been weary of wildcard and muti-San certs that preclude a > certificate for each server. In our case, we have got a multi-san cert for > each expressway E (and C for that matter) which includes the server as the > primary host, and the peer, cluster name and domain as a SAN. > > > > I’m lucky that our cert team has got a contract with good inventory, so, a > couple of extra multi-SAN certs isn’t a big deal for us. > > > > At some point, we may consider moving the Expressways to Let’s Encrypt. It’s > the only Cisco collab platform that supports it for now. > > > > > > From: cisco-voip On Behalf Of > Shaihan Jaffrey > Sent: Tuesday, August 2, 2022 4:21 AM > To: Cisco VOIP > Subject: [cisco-voip] expressway E GoDaddy certificate > > > > CAUTION: This email originated from outside of the University of > Guelph. Do not click links or open attachments unless you recognize > the sender and know the content is safe. If in doubt, forward > suspicious emails to ith...@uoguelph.ca > > > > what is the process to renew Public certificate on Expressway E > through > > GoDaddy. > > Is one certificate sufficient for primary and secondary exp-e? > > > > do we have to get certificates based on FQDN? > > > > Regards > > ___ > 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] Expressway warning on unsupported VM configuration
Thanks. Lovely. Just didn't want to open a case with TAC and told they couldn't help because we are "unsupported". From: James Dust Sent: Tuesday, July 26, 2022 7:24 AM To: Matthew Huff ; cisco-voip voyp list Subject: RE: Expressway warning on unsupported VM configuration Hi Matthew, I had the same issue last week, and was told by cisco tac it is a bug and nothing to be concerned about: Bug Search Tool (cisco.com)<https://quickview.cloudapps.cisco.com/quickview/bug/CSCvy24672> Kind regards James From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Matthew Huff Sent: 26 July 2022 12:20 To: cisco-voip voyp list mailto:cisco-voip@puck.nether.net>> Subject: [cisco-voip] Expressway warning on unsupported VM configuration This message originates from outside Charles Stanley. Please do not click links or open attachments unless you know the sender and are confident the content is safe. After upgrading from 14.0.5 to 14.0.8 on my expreway VMs, it now gives a warning about unsupported VM configuration. The only info is it points to the web page given Cisco's VM requirements for UCS and third-party hosts. At least everything I see, we fit those requirements, but I'm betting it some new requirement relating to the generate of Intel processors. Does anyone know how to get more info about what it's warning about? Consider the environment - Think before you print The contents of this email are confidential to the intended recipient and may not be disclosed. Although it is believed that this email and any attachments are virus free, it is the responsibility of the recipient to confirm this. You are advised that urgent, time-sensitive communications should not be sent by email. We hereby give you notice that a delivery receipt does not constitute acknowledgement or receipt by the intended recipient(s). Details of Charles Stanley group companies and their regulators (where applicable), can be found at this URL http://www.charles-stanley.co.uk/contact-us/disclosure/ ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Expressway warning on unsupported VM configuration
After upgrading from 14.0.5 to 14.0.8 on my expreway VMs, it now gives a warning about unsupported VM configuration. The only info is it points to the web page given Cisco's VM requirements for UCS and third-party hosts. At least everything I see, we fit those requirements, but I'm betting it some new requirement relating to the generate of Intel processors. Does anyone know how to get more info about what it's warning about? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
Yes, they want 6 boxes for redundancy. Two for expressway-e, two for CUCM, two for expressway-c. /Boggle Even then, that doesn't provide 100% redundancy. We want to place our CUCM and expressways at different datacenters connected by a 10GB wan. If we were to loose the WAN, we would still fail with MRA since we would lose both ESXi hosts. -Original Message- From: Lelio Fulgenzi Sent: Tuesday, June 21, 2022 4:30 PM To: Matthew Huff ; Adam Pawlowski ; cisco-voip voyp list Subject: RE: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect I was miffed when they said no real MRA redundancy until you upgrade to v14. But now, hearing this, man, what a disappointment. I had a similar discussion with the Expressway folks and ESXi compatibility/testing and they were like, yeah, you should probably have separate UCS boxes for Expressways different than your CUCMs. And I was all, "wait, what?" They want us to run a completely separate ESXi box with only an E or a C on it to get full MRA redundancy? What a let down. ☹ -Original Message- From: cisco-voip On Behalf Of Matthew Huff Sent: Tuesday, June 21, 2022 1:37 PM To: Adam Pawlowski ; cisco-voip voyp list Subject: Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca Yes, that sounds almost exactly what we are experiencing. I think it's a design defect with the MRA architecture with the end device not downloading/retrying with the full environment. It's the same issue we have with partial registrations. We have a number of shared SIP lines (think SALES line) that can silently fail on phone. It will try for a few minutes, but give up after that. The user doesn't know that the shared SIP line is disconnected, they just don't get calls on it. We had to add a complex SNMP monitoring so that we can be alerted when this happen and remotely reset the phones. Cisco TAC is aware of this issue and also told us it's "working as intended". We had a sales trader lose about $10k of commission because he missed a call, and he was not a happy camper. -Original Message- From: cisco-voip On Behalf Of Adam Pawlowski Sent: Tuesday, June 21, 2022 1:04 PM To: cisco-voip voyp list Subject: Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect I'm a bit miffed on the need for the extra expressway C. We have very few MRA phones, but hadn't had this type of problem, an expressway is somehow busted and not accepting registrations - did they offer any explanation as to why that piece is needed? The only thing I'd go to look up is how the CM list is being populated, if changing the CM group to bump the shut subscriber down (assuming reg order is sub -> pub), just since that'd come up before. Expressway doesn't seem to be configured to be aware that a UCM has gone away, despite the zone going down, at least for UDS and discovery. I'm sure TAC looked at that though. I have a conversation going with them about this and Jabber SSO for a similar reason, that the device's configuration isn't dynamic to represent the state of the infrastructure, and sometimes they get stuck trying something that won't work and fail despite other components being available to serve them. That probably doesn't help with anything other than to say we're in a similar boat, just with Jabber and MRA. Adam Pawlowski Network Engineer | Network and Communication Services University at Buffalo Information Technology (UBIT) 243 Computing Center, Buffalo, NY 14260 > -Original Message- > From: cisco-voip On Behalf Of > Matthew Huff > Sent: Tuesday, June 21, 2022 12:54 PM > To: Hunter Fuller > Cc: cisco-voip voyp list > Subject: Re: [cisco-voip] [External] Re: MRA failover doesn't work, > Cisco TAC agrees, says it's a documentation defect > > We have no interest in setting up a jabber environment in order to > debug ciscos's issue. > > Yes, every expressway-e knows about all expressway-c, all expressway-c > know about CUCM. Cisco TAC has verified the configuration, logs, and > diagnostic. I've been working with them for 2 months and it's been > escalated to backline-engineering. They looked at the Cisco Phone PRT > logs and confirmed that it's a known limitation, and there is no solution. > > Maybe it's an issue with later versions of CUCM and/or expressway? We > are running the latest including latest phone firmware. > > Failover works great except in one scenario where both the CUCM > subscriber and the expressway-c that reside
Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
Yes, that sounds almost exactly what we are experiencing. I think it's a design defect with the MRA architecture with the end device not downloading/retrying with the full environment. It's the same issue we have with partial registrations. We have a number of shared SIP lines (think SALES line) that can silently fail on phone. It will try for a few minutes, but give up after that. The user doesn't know that the shared SIP line is disconnected, they just don't get calls on it. We had to add a complex SNMP monitoring so that we can be alerted when this happen and remotely reset the phones. Cisco TAC is aware of this issue and also told us it's "working as intended". We had a sales trader lose about $10k of commission because he missed a call, and he was not a happy camper. -Original Message- From: cisco-voip On Behalf Of Adam Pawlowski Sent: Tuesday, June 21, 2022 1:04 PM To: cisco-voip voyp list Subject: Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect I'm a bit miffed on the need for the extra expressway C. We have very few MRA phones, but hadn't had this type of problem, an expressway is somehow busted and not accepting registrations - did they offer any explanation as to why that piece is needed? The only thing I'd go to look up is how the CM list is being populated, if changing the CM group to bump the shut subscriber down (assuming reg order is sub -> pub), just since that'd come up before. Expressway doesn't seem to be configured to be aware that a UCM has gone away, despite the zone going down, at least for UDS and discovery. I'm sure TAC looked at that though. I have a conversation going with them about this and Jabber SSO for a similar reason, that the device's configuration isn't dynamic to represent the state of the infrastructure, and sometimes they get stuck trying something that won't work and fail despite other components being available to serve them. That probably doesn't help with anything other than to say we're in a similar boat, just with Jabber and MRA. Adam Pawlowski Network Engineer | Network and Communication Services University at Buffalo Information Technology (UBIT) 243 Computing Center, Buffalo, NY 14260 > -Original Message- > From: cisco-voip On Behalf Of > Matthew Huff > Sent: Tuesday, June 21, 2022 12:54 PM > To: Hunter Fuller > Cc: cisco-voip voyp list > Subject: Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC > agrees, says it's a documentation defect > > We have no interest in setting up a jabber environment in order to debug > ciscos's issue. > > Yes, every expressway-e knows about all expressway-c, all expressway-c > know about CUCM. Cisco TAC has verified the configuration, logs, and > diagnostic. I've been working with them for 2 months and it's been escalated > to backline-engineering. They looked at the Cisco Phone PRT logs and > confirmed that it's a known limitation, and there is no solution. > > Maybe it's an issue with later versions of CUCM and/or expressway? We are > running the latest including latest phone firmware. > > Failover works great except in one scenario where both the CUCM subscriber > and the expressway-c that reside on the same machine are both shut down. > Brining either one up, and the phone registers. > > > -----Original Message- > From: Hunter Fuller > Sent: Tuesday, June 21, 2022 12:41 PM > To: Matthew Huff > Cc: Kent Roberts ; cisco-voip voyp list v...@puck.nether.net> > Subject: Re: [External] Re: [cisco-voip] MRA failover doesn't work, Cisco TAC > agrees, says it's a documentation defect > > It might be worth setting up a Jabber test endpoint just to see. > > Some questions though: > - Does every Expressway-E know about every Expressway-C? > - Does every Expressway-C know about every CUCM? > > I'm trying to figure out what the desired architecture is, and/or how this > problem would happen. > In our environment, the above are both true. So the loss of any number of > anything, should not result in failover issues - and that is the behavior we > have seen (we have shut down entire sites due to maintenance, power > failure, etc. and failover worked). > In fact, we have found MRA phones to be great at failover in this way (our > MRA phones are all 8851s). Jabber has been the problem child. > > -- > Hunter Fuller (they) > Router Jockey > VBH M-1C > +1 256 824 5331 > > Office of Information Technology > The University of Alabama in Huntsville > Network Engineering > > On Tue, Jun 21, 2022 at 9:13 AM Matthew Huff wrote: > > > > We don’t use Jabber nor Webex. > > > > > > > > Cisco TAC has been escalated and they have been working on this for over > 2 mon
Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
The only issue with them being on the same server is that both have to be shutdown to do hardware maintenance or VMWare patching. Their solution is to buy another vmware server and separate the expressway and CUCM onto separate servers so that they can be shut down separately. I guess they expect 1 ESXi host = 1 VM. /boggle. I don't think it's the fact that they are on the same host, I think the phone only downloads limited knowledge of the environment and when there is "enough" of a failure, it doesn't know enough to contact the other servers. It looks like a design defect on the phone firmware/MRA not necessarily CUCM. -Original Message- From: Hunter Fuller Sent: Tuesday, June 21, 2022 1:05 PM To: Matthew Huff Cc: Kent Roberts ; cisco-voip voyp list Subject: Re: [External] Re: [cisco-voip] MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect What Cisco is saying doesn't make sense to me. In a scenario like you have described, where every server knows about every other server, what is the difference in the Expressway-C and CUCM being on the same machine? Since they are all "equals" in this configuration, it should not matter where the Expressway-C and CUCM are. I guess what I'm suggesting is, is it possible that the failure of ANY CUCM and ANY Exp-C at the same time, is causing this issue? Another test could be to shut down and move the Exp-C VMs between hosts. (Not using vMotion obviously) If this resolves the issue (e.g., CUCM Subscriber and Expressway-C-1 are on the same host now, and killing that host does NOT result in an outage anymore), then we will learn that there is some specific thing lurking in the config between specific CUCM and specific Exp-C that is causing the issue. If it does not resolve the issue, then you can test manually powering off CUCM and Exp-C but on different hosts. This would test whether it is just an issue with simultaneous failure of ANY CUCM+Exp-C at once. I hope what I'm saying makes sense. The UC architecture does not "know" about what VM host the apps live on. So there should be no special relationship between VMs on the same host. That is why it smells like something else is going on (despite what Cisco says). -- Hunter Fuller (they) Router Jockey VBH M-1C +1 256 824 5331 Office of Information Technology The University of Alabama in Huntsville Network Engineering On Tue, Jun 21, 2022 at 11:54 AM Matthew Huff wrote: > > We have no interest in setting up a jabber environment in order to debug > ciscos's issue. > > Yes, every expressway-e knows about all expressway-c, all expressway-c know > about CUCM. Cisco TAC has verified the configuration, logs, and diagnostic. > I've been working with them for 2 months and it's been escalated to > backline-engineering. They looked at the Cisco Phone PRT logs and confirmed > that it's a known limitation, and there is no solution. > > Maybe it's an issue with later versions of CUCM and/or expressway? We are > running the latest including latest phone firmware. > > Failover works great except in one scenario where both the CUCM subscriber > and the expressway-c that reside on the same machine are both shut down. > Brining either one up, and the phone registers. > > > -Original Message- > From: Hunter Fuller > Sent: Tuesday, June 21, 2022 12:41 PM > To: Matthew Huff > Cc: Kent Roberts ; cisco-voip voyp list > > Subject: Re: [External] Re: [cisco-voip] MRA failover doesn't work, > Cisco TAC agrees, says it's a documentation defect > > It might be worth setting up a Jabber test endpoint just to see. > > Some questions though: > - Does every Expressway-E know about every Expressway-C? > - Does every Expressway-C know about every CUCM? > > I'm trying to figure out what the desired architecture is, and/or how this > problem would happen. > In our environment, the above are both true. So the loss of any number of > anything, should not result in failover issues - and that is the behavior we > have seen (we have shut down entire sites due to maintenance, power failure, > etc. and failover worked). > In fact, we have found MRA phones to be great at failover in this way (our > MRA phones are all 8851s). Jabber has been the problem child. > > -- > Hunter Fuller (they) > Router Jockey > VBH M-1C > +1 256 824 5331 > > Office of Information Technology > The University of Alabama in Huntsville Network Engineering > > On Tue, Jun 21, 2022 at 9:13 AM Matthew Huff wrote: > > > > We don’t use Jabber nor Webex. > > > > > > > > Cisco TAC has been escalated and they have been working on this for over 2 > > months. I have sent repeated expressway and PRT logs from the phone. After > > working with Cisco engineering, the
Re: [cisco-voip] [External] Re: MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
We have no interest in setting up a jabber environment in order to debug ciscos's issue. Yes, every expressway-e knows about all expressway-c, all expressway-c know about CUCM. Cisco TAC has verified the configuration, logs, and diagnostic. I've been working with them for 2 months and it's been escalated to backline-engineering. They looked at the Cisco Phone PRT logs and confirmed that it's a known limitation, and there is no solution. Maybe it's an issue with later versions of CUCM and/or expressway? We are running the latest including latest phone firmware. Failover works great except in one scenario where both the CUCM subscriber and the expressway-c that reside on the same machine are both shut down. Brining either one up, and the phone registers. -Original Message- From: Hunter Fuller Sent: Tuesday, June 21, 2022 12:41 PM To: Matthew Huff Cc: Kent Roberts ; cisco-voip voyp list Subject: Re: [External] Re: [cisco-voip] MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect It might be worth setting up a Jabber test endpoint just to see. Some questions though: - Does every Expressway-E know about every Expressway-C? - Does every Expressway-C know about every CUCM? I'm trying to figure out what the desired architecture is, and/or how this problem would happen. In our environment, the above are both true. So the loss of any number of anything, should not result in failover issues - and that is the behavior we have seen (we have shut down entire sites due to maintenance, power failure, etc. and failover worked). In fact, we have found MRA phones to be great at failover in this way (our MRA phones are all 8851s). Jabber has been the problem child. -- Hunter Fuller (they) Router Jockey VBH M-1C +1 256 824 5331 Office of Information Technology The University of Alabama in Huntsville Network Engineering On Tue, Jun 21, 2022 at 9:13 AM Matthew Huff wrote: > > We don’t use Jabber nor Webex. > > > > Cisco TAC has been escalated and they have been working on this for over 2 > months. I have sent repeated expressway and PRT logs from the phone. After > working with Cisco engineering, the claim it is “working as intended” and > plan on updating the documentation to reflect the limitation that if you > loose both the subscriber and redundant expressway-C server, failover won’t > happen. > > > > I’d love to be proven wrong since we may have to completely replace our > solution. > > > > > > From: Kent Roberts > Sent: Tuesday, June 21, 2022 10:09 AM > To: Matthew Huff > Cc: cisco-voip voyp list > Subject: Re: [cisco-voip] MRA failover doesn't work, Cisco TAC agrees, > says it's a documentation defect > > > > This sound more like a config issue… > > > > Have run into issues where expressways go stupid when boxes go offline > > As for it being the phones 88xx. Does the same happen with jabber or webex? > If it does i’d requeue the case…. > > > > Kent > > > > On Jun 21, 2022, at 07:47, Matthew Huff wrote: > > > > We have a fairly common and standard deployment for our MRA solution. > All are running CUCM 14+, latest Expressway, etc… > > > > Vmware server 1 (jn DMZ) > > ExpressWay-E-1 > > > > Vmware server 2 (in DMZ) > >ExpressWay-E-2 > > > > Vmware Server 3 (In Core) > > CUCM Publisher > >Expressway-C-1 > > > > VMWare Server 4( In Core) > >CUCM Subscriber > >Expressway-C-2 > > > > > > If ether Expreway-E VMs fail, redundancy works fine If either CUCM > fails, redundancy works fine If either Expressway-C VMs fail, > redundancy works fine If VMWare Server 4 fails (say during patching, > hardware maintenance or hardware failure), redundancy fails. Remote phones > un-register and never register no matter what is done. If either CUCM > Subscriber or Expressway-C-2 is brought back online, phones register. > > > > Cisco TAC claims that this is a limitation of our Cisco 88xx SIP MRA phones > and is not solvable unless we purchase two new vmware servers and split the > CUCM and Expressway-C into separate servers so they both won’t go down at > once. Sinc VMWare Server 3 & 4 are at different locations, vMotion isn’t an > option since there is no shared storage. > > > > Anyone run into this or have any suggestions? We have engaged our VAR and > cisco rep and may have to replace our phone system since we are all working > from home and MRA support including redundancy is critical to us. > > > > > > > > ___ > 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
Re: [cisco-voip] MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
We don’t use Jabber nor Webex. Cisco TAC has been escalated and they have been working on this for over 2 months. I have sent repeated expressway and PRT logs from the phone. After working with Cisco engineering, the claim it is “working as intended” and plan on updating the documentation to reflect the limitation that if you loose both the subscriber and redundant expressway-C server, failover won’t happen. I’d love to be proven wrong since we may have to completely replace our solution. From: Kent Roberts Sent: Tuesday, June 21, 2022 10:09 AM To: Matthew Huff Cc: cisco-voip voyp list Subject: Re: [cisco-voip] MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect This sound more like a config issue… Have run into issues where expressways go stupid when boxes go offline As for it being the phones 88xx. Does the same happen with jabber or webex? If it does i’d requeue the case…. Kent On Jun 21, 2022, at 07:47, Matthew Huff mailto:mh...@ox.com>> wrote: We have a fairly common and standard deployment for our MRA solution. All are running CUCM 14+, latest Expressway, etc… Vmware server 1 (jn DMZ) ExpressWay-E-1 Vmware server 2 (in DMZ) ExpressWay-E-2 Vmware Server 3 (In Core) CUCM Publisher Expressway-C-1 VMWare Server 4( In Core) CUCM Subscriber Expressway-C-2 1. If ether Expreway-E VMs fail, redundancy works fine 2. If either CUCM fails, redundancy works fine 3. If either Expressway-C VMs fail, redundancy works fine 4. If VMWare Server 4 fails (say during patching, hardware maintenance or hardware failure), redundancy fails. Remote phones un-register and never register no matter what is done. If either CUCM Subscriber or Expressway-C-2 is brought back online, phones register. Cisco TAC claims that this is a limitation of our Cisco 88xx SIP MRA phones and is not solvable unless we purchase two new vmware servers and split the CUCM and Expressway-C into separate servers so they both won’t go down at once. Sinc VMWare Server 3 & 4 are at different locations, vMotion isn’t an option since there is no shared storage. Anyone run into this or have any suggestions? We have engaged our VAR and cisco rep and may have to replace our phone system since we are all working from home and MRA support including redundancy is critical to us. ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] MRA failover doesn't work, Cisco TAC agrees, says it's a documentation defect
We have a fairly common and standard deployment for our MRA solution. All are running CUCM 14+, latest Expressway, etc... Vmware server 1 (jn DMZ) ExpressWay-E-1 Vmware server 2 (in DMZ) ExpressWay-E-2 Vmware Server 3 (In Core) CUCM Publisher Expressway-C-1 VMWare Server 4( In Core) CUCM Subscriber Expressway-C-2 1. If ether Expreway-E VMs fail, redundancy works fine 2. If either CUCM fails, redundancy works fine 3. If either Expressway-C VMs fail, redundancy works fine 4. If VMWare Server 4 fails (say during patching, hardware maintenance or hardware failure), redundancy fails. Remote phones un-register and never register no matter what is done. If either CUCM Subscriber or Expressway-C-2 is brought back online, phones register. Cisco TAC claims that this is a limitation of our Cisco 88xx SIP MRA phones and is not solvable unless we purchase two new vmware servers and split the CUCM and Expressway-C into separate servers so they both won't go down at once. Sinc VMWare Server 3 & 4 are at different locations, vMotion isn't an option since there is no shared storage. Anyone run into this or have any suggestions? We have engaged our VAR and cisco rep and may have to replace our phone system since we are all working from home and MRA support including redundancy is critical to us. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES
Just figured that out. D’oh. Looks to be easy now. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: russon81 Sent: Thursday, March 10, 2022 10:54 AM To: Matthew Huff ; NateCCIE Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES I think if you have the SIP trunk assigned to a route pattern, it won't show up as an option in the route groups. You can definitely assign them to a RG though. Try creating the RG, point your pattern to that, then you should be able to assign the trunk to the RG. Sent from my T-Mobile 5G Device Original message ---- From: Matthew Huff mailto:mh...@ox.com>> Date: 3/10/22 7:46 AM (GMT-08:00) To: NateCCIE mailto:natec...@gmail.com>> Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES That was my initial thought as well, but I could only see how to setup a route group with H.323/QSIG,MGCP, etc, not SIP. I’ll look again. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: NateCCIE mailto:natec...@gmail.com>> Sent: Thursday, March 10, 2022 10:29 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES Split it to two different trunks and add them both to a route group that is configured for round robin to the route list. On Mar 10, 2022, at 8:13 AM, Matthew Huff mailto:mh...@ox.com>> wrote: We have two cisco ISR 4331 Cube gateways in two different locations. I want to be able to route calls to both devices (preferably round-robin). I have the router pattern going to a trunk with both cubes defined (with sip options keep-alive configured). The issue we are having is that if the call is made to CUBE1 and the associated outbound dial-peer in in busyout, the CUBE returns 503 Service unavailable and CUCM doesn’t try CUBE2. What am I missing? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] CUCM SIP trunk redundancy with multiple CUBES
That was my initial thought as well, but I could only see how to setup a route group with H.323/QSIG,MGCP, etc, not SIP. I’ll look again. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: NateCCIE Sent: Thursday, March 10, 2022 10:29 AM To: Matthew Huff Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES Split it to two different trunks and add them both to a route group that is configured for round robin to the route list. On Mar 10, 2022, at 8:13 AM, Matthew Huff mailto:mh...@ox.com>> wrote: We have two cisco ISR 4331 Cube gateways in two different locations. I want to be able to route calls to both devices (preferably round-robin). I have the router pattern going to a trunk with both cubes defined (with sip options keep-alive configured). The issue we are having is that if the call is made to CUBE1 and the associated outbound dial-peer in in busyout, the CUBE returns 503 Service unavailable and CUCM doesn’t try CUBE2. What am I missing? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] CUCM SIP trunk redundancy with multiple CUBES
We have two cisco ISR 4331 Cube gateways in two different locations. I want to be able to route calls to both devices (preferably round-robin). I have the router pattern going to a trunk with both cubes defined (with sip options keep-alive configured). The issue we are having is that if the call is made to CUBE1 and the associated outbound dial-peer in in busyout, the CUBE returns 503 Service unavailable and CUCM doesn't try CUBE2. What am I missing? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SIP to iTSP best practices
We have a few fax phone numbers that have been used for 20+ years. They are in corporate documents and regulatory filings. Since there is a just a few, I bought a couple of ATA and can play around with them and move them over time. For personal faxing, we already use e-faxing. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Myron Young Sent: Friday, February 11, 2022 5:31 PM To: Matthew Huff Cc: Kent Roberts ; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] SIP to iTSP best practices If you can just go with E-faxing, do that because it will save you lots of headaches as well. On Feb 11, 2022, at 12:43 PM, Matthew Huff mailto:mh...@ox.com>> wrote: Thanks. Our new SIP voice gateway is separate and not in production so I have plenty of freedom to play. We have copper based FAX lines, not going over our PRI currently. This is something we are looking into though after this conversion is done. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Kent Roberts mailto:dvx...@gmail.com>> Sent: Friday, February 11, 2022 12:14 PM To: Matthew Huff mailto:mh...@ox.com>>; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] SIP to iTSP best practices Oh yeah.. one more thing... Test faxing a fax test is a min of 10 pages, inbound call and out don't just do a page and say your good. Check T38 if your using it... if you have to fail back because of T38 non-compliant, is G711 working? Does your faxing software do/support switchback to 711 if T38 doesn't setup. If you have a fax machine on a ATA or whater, test to it as well. Isn't fax dead yet? :) good luck with your go live. On 2/11/22 8:52 AM, Matthew Huff wrote: Thanks for the recommendations. I have a lot to dig into. Question about the video disable. We have no video hardware, so think it would be good to disable it before we go live. What’s the best way to disable it globally? Is it Voice service voip Sip Audio forced ? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Kent Roberts <mailto:dvx...@gmail.com> Sent: Thursday, February 10, 2022 6:14 PM To: Matthew Huff <mailto:mh...@ox.com>; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] SIP to iTSP best practices I was part of the team that starting a large scale sip migration almost 10 years ago. Have moved thousand's of DID since then. Run multiple gig circuits into the cube. Recommendations: on the link to your provider, use address outside of the route able block for your company. (say you use 10.x.x.x then use 172.16 or 192.168) If you can, don't route the itsp connections on your company network, go direct to the routers supporting those links. (BGP peers I would guess depending on carrier/build) If you can use a dedicated router, unless is a small site This is important if you wind up doing any kind of call recording, or if you have to enable debugs during the day. Use dedicated dial peers setup exactly for each itsp SBC link for in and one for out. Use something like the "voice class uri trunk(x) sip" or equivalent to bind to the dial peers for each SBC. This will help if you have to add additional carriers, or say acquire a company, or need to do special routing... use full E164 to and from the carrier, they may only want to do 10 digit in/out, but that is easy enough. (uri trunkx will help here, as the inbound number will be at the cube, then you can route to cucm with outbound dial peer) From your CUCM still send the 9 or 8 or whatever for outbound, then strip on match in the dialpeer to Itsp. This will keep call looping etc. define your voice class codecs on the dialpeers... don't just assume it will take the default, or work as you want without it. if the cube will never see VIDEO, disable the options. The cube software likes to release bugs that cause the cube to go south with video errors. Depending on your carrier, you may need to force G729 or G711 first, even if its not your preferred codec, have seen were the SBC will not negotiate a call, if the codecs aren't
Re: [cisco-voip] SIP to iTSP best practices
Thanks. Our new SIP voice gateway is separate and not in production so I have plenty of freedom to play. We have copper based FAX lines, not going over our PRI currently. This is something we are looking into though after this conversion is done. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Kent Roberts Sent: Friday, February 11, 2022 12:14 PM To: Matthew Huff ; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] SIP to iTSP best practices Oh yeah.. one more thing... Test faxing a fax test is a min of 10 pages, inbound call and out don't just do a page and say your good. Check T38 if your using it... if you have to fail back because of T38 non-compliant, is G711 working? Does your faxing software do/support switchback to 711 if T38 doesn't setup. If you have a fax machine on a ATA or whater, test to it as well. Isn't fax dead yet? :) good luck with your go live. On 2/11/22 8:52 AM, Matthew Huff wrote: Thanks for the recommendations. I have a lot to dig into. Question about the video disable. We have no video hardware, so think it would be good to disable it before we go live. What’s the best way to disable it globally? Is it Voice service voip Sip Audio forced ? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Kent Roberts <mailto:dvx...@gmail.com> Sent: Thursday, February 10, 2022 6:14 PM To: Matthew Huff <mailto:mh...@ox.com>; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] SIP to iTSP best practices I was part of the team that starting a large scale sip migration almost 10 years ago. Have moved thousand's of DID since then. Run multiple gig circuits into the cube. Recommendations: on the link to your provider, use address outside of the route able block for your company. (say you use 10.x.x.x then use 172.16 or 192.168) If you can, don't route the itsp connections on your company network, go direct to the routers supporting those links. (BGP peers I would guess depending on carrier/build) If you can use a dedicated router, unless is a small site This is important if you wind up doing any kind of call recording, or if you have to enable debugs during the day. Use dedicated dial peers setup exactly for each itsp SBC link for in and one for out. Use something like the "voice class uri trunk(x) sip" or equivalent to bind to the dial peers for each SBC. This will help if you have to add additional carriers, or say acquire a company, or need to do special routing... use full E164 to and from the carrier, they may only want to do 10 digit in/out, but that is easy enough. (uri trunkx will help here, as the inbound number will be at the cube, then you can route to cucm with outbound dial peer) From your CUCM still send the 9 or 8 or whatever for outbound, then strip on match in the dialpeer to Itsp. This will keep call looping etc. define your voice class codecs on the dialpeers... don't just assume it will take the default, or work as you want without it. if the cube will never see VIDEO, disable the options. The cube software likes to release bugs that cause the cube to go south with video errors. Depending on your carrier, you may need to force G729 or G711 first, even if its not your preferred codec, have seen were the SBC will not negotiate a call, if the codecs aren't in the order the carriers SBC wants. do not assume the carriers network will normalize the calls. For instance, if the destination is on the same carrier, its a direct ip route via the SBC. If that end side can't accept say G729 (cheaper sbc) the call will just fail. NEVER user debug ccsip all debug CCSIP messages is safer, and unless your cube is peeked, it won't add to much cpu. make sure your CPU never exceeds 80% at the max possible peek of routing. Check how the calls work with MOH. Inbound and out. make sure 2 way audio remains after the on hold event.. Do you need to force early offer? (yes sounds silly, but have run into issues where some phones had no audio unless this was set) Ask your carrier, how they handle TFNs outbound, if you pass the ANI from a 3rd party. (this is all billing stuff to the TFN owner) Some may allow calls to process not caring what the number is. Some may allow you to provide a alternate billing number. Some will just 603 decline the call if the ANI isn't in your numbe
Re: [cisco-voip] SIP to iTSP best practices
Thanks for the recommendations. I have a lot to dig into. Question about the video disable. We have no video hardware, so think it would be good to disable it before we go live. What’s the best way to disable it globally? Is it Voice service voip Sip Audio forced ? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Kent Roberts Sent: Thursday, February 10, 2022 6:14 PM To: Matthew Huff ; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] SIP to iTSP best practices I was part of the team that starting a large scale sip migration almost 10 years ago. Have moved thousand's of DID since then. Run multiple gig circuits into the cube. Recommendations: on the link to your provider, use address outside of the route able block for your company. (say you use 10.x.x.x then use 172.16 or 192.168) If you can, don't route the itsp connections on your company network, go direct to the routers supporting those links. (BGP peers I would guess depending on carrier/build) If you can use a dedicated router, unless is a small site This is important if you wind up doing any kind of call recording, or if you have to enable debugs during the day. Use dedicated dial peers setup exactly for each itsp SBC link for in and one for out. Use something like the "voice class uri trunk(x) sip" or equivalent to bind to the dial peers for each SBC. This will help if you have to add additional carriers, or say acquire a company, or need to do special routing... use full E164 to and from the carrier, they may only want to do 10 digit in/out, but that is easy enough. (uri trunkx will help here, as the inbound number will be at the cube, then you can route to cucm with outbound dial peer) From your CUCM still send the 9 or 8 or whatever for outbound, then strip on match in the dialpeer to Itsp. This will keep call looping etc. define your voice class codecs on the dialpeers... don't just assume it will take the default, or work as you want without it. if the cube will never see VIDEO, disable the options. The cube software likes to release bugs that cause the cube to go south with video errors. Depending on your carrier, you may need to force G729 or G711 first, even if its not your preferred codec, have seen were the SBC will not negotiate a call, if the codecs aren't in the order the carriers SBC wants. do not assume the carriers network will normalize the calls. For instance, if the destination is on the same carrier, its a direct ip route via the SBC. If that end side can't accept say G729 (cheaper sbc) the call will just fail. NEVER user debug ccsip all debug CCSIP messages is safer, and unless your cube is peeked, it won't add to much cpu. make sure your CPU never exceeds 80% at the max possible peek of routing. Check how the calls work with MOH. Inbound and out. make sure 2 way audio remains after the on hold event.. Do you need to force early offer? (yes sounds silly, but have run into issues where some phones had no audio unless this was set) Ask your carrier, how they handle TFNs outbound, if you pass the ANI from a 3rd party. (this is all billing stuff to the TFN owner) Some may allow calls to process not caring what the number is. Some may allow you to provide a alternate billing number. Some will just 603 decline the call if the ANI isn't in your number poll assigned to you. with a 603 the cube will try the next dial peer so you can add a header to re-write this with your number. Diversion headers exist, however most carriers pass them through to the destination, and IVRs or Voice Mail systems on the far side will try to process that information, and do unexpected things. (the party your calling doesn't exist for example.) define the default sip control/media source interface, this will be your destination from cucm. The URI trucks will define the sip control/media on the ITSP side. If you use firewalls any where in your company, that will pass voip... Set the rtp-port range on the cube match the smaller range of what your going to use. (say the old days 16384-32767) don't assume the firewall will pass all the UDP ports by default. speaking of firewalls, check, double check, and triple check, then do your own research if you will use them, when it comes to SIP inspection. Some FW's have options that need to be tweeked and defined, for the SIP port. (this may control anything from timeouts, which media ports engage)This is especially true with expressway in the DMZ. It might be safer to not use sip inspection and just pass the port. But for some FWs this is not true. define the FAX-relay, rats and proto
Re: [cisco-voip] SIP to iTSP best practices
That’s the plan. Slow migration. Setup test inbound DID and use a different # to dial outbound for testing via SIP. Once everything is confirmed then migrate routing and port inbound DIDs. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Lelio Fulgenzi Sent: Thursday, February 10, 2022 1:43 PM To: Matthew Huff Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] SIP to iTSP best practices The only thing I’ve heard and would probably keep is a couple of PRIs for facing until you’ve ironed out everything else. Sent from my iPhone On Feb 10, 2022, at 11:14 AM, Matthew Huff mailto:mh...@ox.com>> wrote: CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca<mailto:ith...@uoguelph.ca> We are in the process of migrating for a legacy PTSN voice gateway (PRI) to a new CUBE based SIP connection to a iTSP connected via a private metro ethernet (not Internet based). Does anyone have a good source for recipes / dial-plans recommendations / best practices for this? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] SIP to iTSP best practices
We are in the process of migrating for a legacy PTSN voice gateway (PRI) to a new CUBE based SIP connection to a iTSP connected via a private metro ethernet (not Internet based). Does anyone have a good source for recipes / dial-plans recommendations / best practices for this? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] [External] Re: Small business E911 solution
We have a SIP provider. The issue is that the user requires the same Prime DN no matter their location. We have a user, for example, at our NY office, he has a two different home offices in Massachusetts and one in Idaho. Our SIP provider in order to route E911 requires each phone to an unique DID, which our users will not accept. What is the easiest solution with on-premise CUCM that allows me to satisfy E911 regulations, especially the ones that go into effect in January, without having to have everyone have an unique prime-dn at each location. Is Cisco Emergency Responder the only choice? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Johnson, Tim Sent: Thursday, December 9, 2021 2:20 PM To: Matthew Huff Cc: cisco-voip@puck.nether.net Subject: Re: [External] Re: [cisco-voip] Small business E911 solution Maybe Intrado? Not sure their minimum requirements but I know we started with them when they were West, with only a couple hundred DIDs. On Dec 9, 2021 2:16 PM, Matthew Huff mailto:mh...@ox.com>> wrote: No, hosted solution isn’t an option as we have a number of custom solutions like ring downs, etc… We already have CUCM and Expressway working fine, I just need directions on the simplest solution for E911 for MRA workers. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Matthew Loraditch mailto:mloradi...@heliontechnologies.com>> Sent: Thursday, December 9, 2021 2:00 PM To: Matthew Huff mailto:mh...@ox.com>>; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: RE: Small business E911 solution I’m very curious if you find something. I’m not aware of anything cost effective at your size. RedSky’s minimum purchase for a CUCM based system is 12-14k. Have you looked at moving to a hosted phone system? Almost every vendor I’m aware of includes E911 therein Matthew Loraditch Sr. Network Engineer (He/Him/His) p: 443.541.1518 w: www.heliontechnologies.com<http://www.heliontechnologies.com/> | e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com> [Helion Technologies]<http://www.heliontechnologies.com/> [Facebook]<https://facebook.com/heliontech> [Twitter]<https://twitter.com/heliontech> [LinkedIn]<https://www.linkedin.com/company/helion-technologies> From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Matthew Huff Sent: Thursday, December 9, 2021 1:34 PM To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: [cisco-voip] Small business E911 solution [EXTERNAL] We are in the process of moving from legacy ISDN PRI for inbound/outbound dialing to SIP, and E911 has hit us in the face. We have less than 50 users, where > 90% currently are working from home. They have the same prime dn for both the office phone and their home phone. We have users that have phones in 3-4 locations including in multiple states. What is the simplest solution to setup and maintain that doesn’t require a user to have a separate DID in each location? Cisco Emergency Responder looks like major overkill. Our environment is: CUCM 14.x Cisco Expressway 14.x for MRA Cisco 8861 SIP phones (both at home and at work). Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Small business E911 solution
No, hosted solution isn’t an option as we have a number of custom solutions like ring downs, etc… We already have CUCM and Expressway working fine, I just need directions on the simplest solution for E911 for MRA workers. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Matthew Loraditch Sent: Thursday, December 9, 2021 2:00 PM To: Matthew Huff ; cisco-voip@puck.nether.net Subject: RE: Small business E911 solution I’m very curious if you find something. I’m not aware of anything cost effective at your size. RedSky’s minimum purchase for a CUCM based system is 12-14k. Have you looked at moving to a hosted phone system? Almost every vendor I’m aware of includes E911 therein Matthew Loraditch Sr. Network Engineer (He/Him/His) p: 443.541.1518 w: www.heliontechnologies.com<http://www.heliontechnologies.com/> | e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com> [Helion Technologies]<http://www.heliontechnologies.com/> [Facebook]<https://facebook.com/heliontech> [Twitter]<https://twitter.com/heliontech> [LinkedIn]<https://www.linkedin.com/company/helion-technologies> From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Matthew Huff Sent: Thursday, December 9, 2021 1:34 PM To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: [cisco-voip] Small business E911 solution [EXTERNAL] We are in the process of moving from legacy ISDN PRI for inbound/outbound dialing to SIP, and E911 has hit us in the face. We have less than 50 users, where > 90% currently are working from home. They have the same prime dn for both the office phone and their home phone. We have users that have phones in 3-4 locations including in multiple states. What is the simplest solution to setup and maintain that doesn’t require a user to have a separate DID in each location? Cisco Emergency Responder looks like major overkill. Our environment is: CUCM 14.x Cisco Expressway 14.x for MRA Cisco 8861 SIP phones (both at home and at work). Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Small business E911 solution
We are in the process of moving from legacy ISDN PRI for inbound/outbound dialing to SIP, and E911 has hit us in the face. We have less than 50 users, where > 90% currently are working from home. They have the same prime dn for both the office phone and their home phone. We have users that have phones in 3-4 locations including in multiple states. What is the simplest solution to setup and maintain that doesn't require a user to have a separate DID in each location? Cisco Emergency Responder looks like major overkill. Our environment is: CUCM 14.x Cisco Expressway 14.x for MRA Cisco 8861 SIP phones (both at home and at work). Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] MRA Onboarding via activation code... phone trust list?
I wouldn’t put a lot of weight in the status on the phone with the TLS error, I’ve seen that with working phones. Do you have the phone MRA domain set? We have a separate device pool for MRA devices so it can set the time from external ntp sources. If the time on the phone is off, the crypto can fail as well. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Jonathan Charles Sent: Thursday, November 11, 2021 11:50 AM To: Matthew Huff Cc: Brian Meade ; cisco-voip voyp list Subject: Re: [cisco-voip] MRA Onboarding via activation code... phone trust list? It is running 12.8... it has been locally reg'd before... On Thu, Nov 11, 2021 at 10:44 AM Matthew Huff mailto:mh...@ox.com>> wrote: In the lab, have you tried setting up the phone without MRA and get the firmware uploaded first? Depending on how old the firmware is, you may have issues with onboarding. Our 8861 wouldn’t onboard until at least 12.5. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Jonathan Charles Sent: Thursday, November 11, 2021 11:10 AM To: Brian Meade mailto:bmead...@vt.edu>> Cc: cisco-voip voyp list mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] MRA Onboarding via activation code... phone trust list? On the phone, we see TLS connection failed... the E's cert is signed by Let's Encrypt... On the Expressway E we see some certificate exchange and then resets in the connection... MRA works fine for Jabber just 8845 Activation Code onboarding is failing... Jonathan On Tue, Nov 9, 2021 at 5:57 PM Brian Meade mailto:bmead...@vt.edu>> wrote: What's the console logs show? The Expressway needs to be signed by one of the trusted CAs listed that are part of the phone firmware. The Expressway cert authenticates the phone with the MIC. Do you have activation code onboarding enabled under the MRA config on the Expressway-C? On Fri, Nov 5, 2021, 5:30 PM Jonathan Charles mailto:jonv...@gmail.com>> wrote: So, I set up activation code MRA for an 8845 (lab first)... Cloud onboarding worked, got an activation code, tried it out... Phone kicks back 'check internet connectivtity' and on the status on the phone says: GDS Handshake Succeeded A TLS connection failed... GDS is Cisco's cloud onboarding thingy I am assuming it didn't like the TLS connection the expressway, but I don't see anything in the Expressway logs... There is a bug and it says we need to load a Hydrant cert back into the trust store... https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt67257?rfs=iqvred But where do we need to load it? Tomcat Trust? On the Expressways? The bug doesn't say... it needs to be pushed to the phone's trust list, how do you do that? Thanks! Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] MRA Onboarding via activation code... phone trust list?
In the lab, have you tried setting up the phone without MRA and get the firmware uploaded first? Depending on how old the firmware is, you may have issues with onboarding. Our 8861 wouldn’t onboard until at least 12.5. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: cisco-voip On Behalf Of Jonathan Charles Sent: Thursday, November 11, 2021 11:10 AM To: Brian Meade Cc: cisco-voip voyp list Subject: Re: [cisco-voip] MRA Onboarding via activation code... phone trust list? On the phone, we see TLS connection failed... the E's cert is signed by Let's Encrypt... On the Expressway E we see some certificate exchange and then resets in the connection... MRA works fine for Jabber just 8845 Activation Code onboarding is failing... Jonathan On Tue, Nov 9, 2021 at 5:57 PM Brian Meade mailto:bmead...@vt.edu>> wrote: What's the console logs show? The Expressway needs to be signed by one of the trusted CAs listed that are part of the phone firmware. The Expressway cert authenticates the phone with the MIC. Do you have activation code onboarding enabled under the MRA config on the Expressway-C? On Fri, Nov 5, 2021, 5:30 PM Jonathan Charles mailto:jonv...@gmail.com>> wrote: So, I set up activation code MRA for an 8845 (lab first)... Cloud onboarding worked, got an activation code, tried it out... Phone kicks back 'check internet connectivtity' and on the status on the phone says: GDS Handshake Succeeded A TLS connection failed... GDS is Cisco's cloud onboarding thingy I am assuming it didn't like the TLS connection the expressway, but I don't see anything in the Expressway logs... There is a bug and it says we need to load a Hydrant cert back into the trust store... https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt67257?rfs=iqvred But where do we need to load it? Tomcat Trust? On the Expressways? The bug doesn't say... it needs to be pushed to the phone's trust list, how do you do that? Thanks! Jonathan ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] MRA phones tuck at registering after internet blip
We are just starting to roll out our Cisco 8800 series phones via MRA, and have run into a small problem. As is common with residential ISPs when they do a firmware update, network maintenance, etc overnight the phones get disconnected and even when the internet comes back, the phones are still at "Registering..." on the phones. If the user notices this and powers cycles the phone it comes right back, but it can cause them to miss calls if they don't notice it. Is there any tunables to address this? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] MRA voice quality issues with Optimum/Altice
A continuing problem we have is voice quality issues with our users on Optimum/Altice. Garbled and dropped speech are common. I've played around with different codes without much success. Opus appears to work as well as any others. We are directly peered with Lightpath, and the traceroutes show a direct path, so it's not an issue with congestion at a peering point. We don't have any issue with Verizon or Spectrum users. Not really looking for any solutions, as it's likely a issue with Optimum, but I was curious if this is a common issue. BTW, don't get me started with the Altice One service. The issues with their router alone is horrid. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Single button ARD
“Ringdowns” are something very common in the trading world. Most traders sit in front of what is called a “Turret” instead of a traditional phone. Turrets in the past were mostly key-switch systems with limited routing capability;i.e, everything is hard-wired. Here is the full picture. Customer Turret Button -à Turret PBX à> IPC T1 dedicated channel ………IPC T1 dedicated channel …..-> IOS Gateway-à CUCM-> Our Phones -> Trader button This is setup this way to bypass PSTN, dialtone, etc.. At the customer site they may have many traders with the same button. On our side we may have many traders with the same button Either side at any time may press the button and the other side immediately rings. Anyone can pickup at that side. Since there is only one channel per ringdown, only one call can be placed per ringdown. I have had this working with SCCP phones on call manager express for many years with this: voice-port 0/3/1:0 define Tx-bits idle define Tx-bits seize define Rx-bits idle define Rx-bits seize no vad no comfort-noise connection plar B100 description StockLoan station-id name StockLoan ephone-dn 32 number B100 label Stock Loan description Stock Loan name Stock Loan trunk B101 ! dial-peer voice 300 pots description StockLoan destination-pattern B101 port 0/3/1:0 I’ll take a look at CTI route points. There are a lot of ways of doing things in CUCM and I’m not familiar with a lot of them, so I don’t know where to even look. So I would use a CTI route point for incoming calls and a SIP Dial Rule for outbound? One question about that is that the documentation for SIP dial rules talk about 79xx phones not 88xx series phones. Does it work with them as well? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: cisco-voip On Behalf Of Austin Williams Sent: Saturday, October 9, 2021 12:07 AM To: Kent Roberts Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Single button ARD This one has me curious now too. Took me a second to understand what you were looking for specifically though. I'd think that the Ringdown to a CTI route point forwarded to the PLAR DN on the phone and then a duplicate on the way back should work. I don't think there is anything stopping a PLAR DN from answering a call but Like Kent just said "Stranger things have happened." I might test this out on Monday with a few test phones I have laying around and see what happens. One question though, Do you also want it to auto answer on each side, Kinda like a push to talk or just cause the phone to ring on the other side? I'm going to experiment with both cause I can see a use for both in my setup. On Fri, Oct 8, 2021 at 9:19 PM Kent Roberts mailto:k...@fredf.org>> wrote: If you put a DN on the source it should be able to ring…. All the guys I work with that have interacted with plars all agree… but stranger things have happened in the new versions of cucm… so… :) On Oct 8, 2021, at 8:05 PM, Lelio Fulgenzi mailto:le...@uoguelph.ca>> wrote: I can’t recall, but are we saying that an extension that is configured as a PLAR isn’t able to receive calls? I could have sworn this was possible. If this is true, wouldn’t this solve the issue? One button to both call out one number and receive calls? And a similar one on another phone? Sent from my iPhone On Oct 8, 2021, at 2:27 PM, Matthew Huff mailto:mh...@ox.com>> wrote: CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca<mailto:ith...@uoguelph.ca> Not really, since it requires acknowledgement. My example only goes so far, in reality, one end of the ARD is a T1 line on a IOS voice gateway that goes to another entity. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com/> ... From: Schlotterer, Tommy mailto:tschlotte...@presidio.com>> Sent: Friday, October 8, 2021 2:03 PM To: Matthew Huff mailto:mh...@ox.com>>; Austin Williams mailto:austinpucknet...@gmail.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: RE: [cisco-voip] Single button ARD Sounds like you are trying to setup an intercom. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/featureConfig/CUCM_BK_C7DC69D3_00_cucm-feature-configuration-guide_115/CUC
Re: [cisco-voip] Single button ARD
Not really, since it requires acknowledgement. My example only goes so far, in reality, one end of the ARD is a T1 line on a IOS voice gateway that goes to another entity. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Schlotterer, Tommy Sent: Friday, October 8, 2021 2:03 PM To: Matthew Huff ; Austin Williams Cc: Cisco VoIP Group Subject: RE: [cisco-voip] Single button ARD Sounds like you are trying to setup an intercom. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/featureConfig/CUCM_BK_C7DC69D3_00_cucm-feature-configuration-guide_115/CUCM_BK_C7DC69D3_00_cucm-feature-configuration-guide_115_chapter_011001.html Tommy Schlotterer | Engineer Presidio | presidio.com<http://www.presidio.com/> 5025 Bradenton Ave, Suite B, Dublin, OH 43017 D: +1.419.214.1415 | C: +1.419.706.0259 | tschlotte...@presidio.com<mailto:tschlotte...@presidio.com> [https://www2.presidio.com/signatures/Cybersecurity-Awareness-Month_Email-Footer_2021.jpg] [https://www2.presidio.com/signatures/Presidio-logo-new.png]<http://www.presidio.com/> From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Matthew Huff Sent: Friday, October 8, 2021 1:26 PM To: Austin Williams mailto:austinpucknet...@gmail.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Single button ARD EXTERNAL EMAIL Here is a full example of what I need. Two users, named Steve & Joe each with a 8861 SIP phone on CUCM 14.x Steve has a prime-dn of 6000 Joe has a prime-dn of 6001 On Steve’s phone he has two extra buttons: Speed Dial, labeled [TO JOE], which dials 7001 Directory Number, labeled [ FROM JOE] which has a DN of 7000 On Joe’s phone, he has two extra buttons: Speed Dial, labeled [TO STEVE], which dials 7000 Directory Number, labeled [ FROM STEVE] which has a DN of 7001 That is easy and works fine. But I need it to be single buttons: On Steve’s phone Labeled [JOE], which dials JOE and receives calls from JOE when he presses his [STEVE] button One Joe’s phone Labeled [STEVE], which dials Steve and receives call from Steve when he presses his [JOE] button I have this working on SCCP phones in Call Mnager Express and is very standard in a trading environment. Anyone have any ideas? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ....... From: Matthew Huff Sent: Friday, October 8, 2021 12:54 PM To: 'Austin Williams' mailto:austinpucknet...@gmail.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: RE: [cisco-voip] Single button ARD That is exactly what I’m not trying to do. Everytime I mention PLAR or ARD those are the articles I’m referred to and are not helpful. That example and others are for courtesy phones, hotlines, etc.. What I need to do is when a dedicated T1 channel on the voice gateway (IOS 2800) goes off-hook it routes to a DN on the phone. When the same DN is picked up, it dials a number that routes back to the same channel on the voice gateway. I can do this with two buttons (one being a speed dial, and the other being a DN), but I need this to be a single button. Someone could have 20 different ring downs on the same phone device. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Austin Williams mailto:austinpucknet...@gmail.com>> Sent: Friday, October 8, 2021 12:49 PM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Single button ARD What are your gateways built with, MGCP, H323, SIP? What version of call manager are you working on? Ringdowns for call managers are built using CSSs, Partitions and a Null Translation pattern. Here is a good article on it: https://www.networkcomputing.com/networking/using-cucm-configure-plar-phones On Fri, Oct 8, 2021 at 11:19 AM Matthew Huff mailto:mh...@ox.com>> wrote: I’m trying to setup a button on people’s phone where they will get inbound calls like a (DN) when voice-port goes off-hook on our gateway (connection plar ). When the button is pressed on any phone, it should do
Re: [cisco-voip] Single button ARD
Here is a full example of what I need. Two users, named Steve & Joe each with a 8861 SIP phone on CUCM 14.x Steve has a prime-dn of 6000 Joe has a prime-dn of 6001 On Steve’s phone he has two extra buttons: Speed Dial, labeled [TO JOE], which dials 7001 Directory Number, labeled [ FROM JOE] which has a DN of 7000 On Joe’s phone, he has two extra buttons: Speed Dial, labeled [TO STEVE], which dials 7000 Directory Number, labeled [ FROM STEVE] which has a DN of 7001 That is easy and works fine. But I need it to be single buttons: On Steve’s phone Labeled [JOE], which dials JOE and receives calls from JOE when he presses his [STEVE] button One Joe’s phone Labeled [STEVE], which dials Steve and receives call from Steve when he presses his [JOE] button I have this working on SCCP phones in Call Mnager Express and is very standard in a trading environment. Anyone have any ideas? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ....... From: Matthew Huff Sent: Friday, October 8, 2021 12:54 PM To: 'Austin Williams' Cc: Cisco VoIP Group Subject: RE: [cisco-voip] Single button ARD That is exactly what I’m not trying to do. Everytime I mention PLAR or ARD those are the articles I’m referred to and are not helpful. That example and others are for courtesy phones, hotlines, etc.. What I need to do is when a dedicated T1 channel on the voice gateway (IOS 2800) goes off-hook it routes to a DN on the phone. When the same DN is picked up, it dials a number that routes back to the same channel on the voice gateway. I can do this with two buttons (one being a speed dial, and the other being a DN), but I need this to be a single button. Someone could have 20 different ring downs on the same phone device. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Austin Williams mailto:austinpucknet...@gmail.com>> Sent: Friday, October 8, 2021 12:49 PM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Single button ARD What are your gateways built with, MGCP, H323, SIP? What version of call manager are you working on? Ringdowns for call managers are built using CSSs, Partitions and a Null Translation pattern. Here is a good article on it: https://www.networkcomputing.com/networking/using-cucm-configure-plar-phones On Fri, Oct 8, 2021 at 11:19 AM Matthew Huff mailto:mh...@ox.com>> wrote: I’m trying to setup a button on people’s phone where they will get inbound calls like a (DN) when voice-port goes off-hook on our gateway (connection plar ). When the button is pressed on any phone, it should do the same thing, take the channel off hook on the voice gateway. All the example’s I’ve seen are for doing PLAR when the phone goes offhook or the “new call button” is pressed, but not a separate line. I’ve done this in the past with CME like: voice-port 0/3/1:0 define Tx-bits idle define Tx-bits seize define Rx-bits idle define Rx-bits seize no vad no comfort-noise connection plar B100 ephone-dn 32 number B100 trunk B101 Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Single button ARD
That is exactly what I’m not trying to do. Everytime I mention PLAR or ARD those are the articles I’m referred to and are not helpful. That example and others are for courtesy phones, hotlines, etc.. What I need to do is when a dedicated T1 channel on the voice gateway (IOS 2800) goes off-hook it routes to a DN on the phone. When the same DN is picked up, it dials a number that routes back to the same channel on the voice gateway. I can do this with two buttons (one being a speed dial, and the other being a DN), but I need this to be a single button. Someone could have 20 different ring downs on the same phone device. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Austin Williams Sent: Friday, October 8, 2021 12:49 PM To: Matthew Huff Cc: Cisco VoIP Group Subject: Re: [cisco-voip] Single button ARD What are your gateways built with, MGCP, H323, SIP? What version of call manager are you working on? Ringdowns for call managers are built using CSSs, Partitions and a Null Translation pattern. Here is a good article on it: https://www.networkcomputing.com/networking/using-cucm-configure-plar-phones On Fri, Oct 8, 2021 at 11:19 AM Matthew Huff mailto:mh...@ox.com>> wrote: I’m trying to setup a button on people’s phone where they will get inbound calls like a (DN) when voice-port goes off-hook on our gateway (connection plar ). When the button is pressed on any phone, it should do the same thing, take the channel off hook on the voice gateway. All the example’s I’ve seen are for doing PLAR when the phone goes offhook or the “new call button” is pressed, but not a separate line. I’ve done this in the past with CME like: voice-port 0/3/1:0 define Tx-bits idle define Tx-bits seize define Rx-bits idle define Rx-bits seize no vad no comfort-noise connection plar B100 ephone-dn 32 number B100 trunk B101 Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Single button ARD
I'm trying to setup a button on people's phone where they will get inbound calls like a (DN) when voice-port goes off-hook on our gateway (connection plar ). When the button is pressed on any phone, it should do the same thing, take the channel off hook on the voice gateway. All the example's I've seen are for doing PLAR when the phone goes offhook or the "new call button" is pressed, but not a separate line. I've done this in the past with CME like: voice-port 0/3/1:0 define Tx-bits idle define Tx-bits seize define Rx-bits idle define Rx-bits seize no vad no comfort-noise connection plar B100 ephone-dn 32 number B100 trunk B101 Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Low space on partitions in CUCM
Thanks, but I believe that suggestion only frees up space on the common partition, not the active/inactive ones. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Jason Aarons Sent: Wednesday, September 29, 2021 11:53 AM To: Matthew Huff Cc: Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM run the free common space. cop once you know you won't be switching version back to old or shutdown and expand vDisk. Delete unused loads. Reduce High and Power Watermark in RTMT. Purge CAR database. On Wed, Sep 29, 2021, 6:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Low space on partitions in CUCM
It was almost certainly done with the smaller partition size originally some years ago. I’m going to look at doing the rebuild, but it will probably be some time before I can justify the maintenance window and resources. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Wes Sisk (wsisk) Sent: Wednesday, September 29, 2021 10:59 AM To: Pete Brown Cc: Lelio Fulgenzi ; Matthew Huff ; Tim Smith ; Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM CSCvt97709 was for systems that installed using an older 80GB OVA and then upgraded. Partition sizes documented in CSCvt97709: TotalFreeUsed Disk/active 14154228K 319736K 13689364K (98%) Disk/inactive 14154228K 828104K 13180996K (95%) Disk/logging49573612K 37501812K9530488K (21%) If a system is still using those partition sizes then a rebuild is worth consideration. Rebuilds are easier with recent automations. If you’re seeing alerts with larger partition sizes then more investigation may be warranted. -Wes On Sep 29, 2021, at 10:21 AM, Pete Brown mailto:j...@chykn.com>> wrote: Yes? From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Lelio Fulgenzi Sent: Wednesday, September 29, 2021 9:09 AM To: Matthew Huff mailto:mh...@ox.com>>; Tim Smith mailto:tim.sm...@enject.com.au>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Oh for the love of pete. Severity: 4 Minor From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net>> On Behalf Of Matthew Huff Sent: Wednesday, September 29, 2021 10:00 AM To: Tim Smith mailto:tim.sm...@enject.com.au>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca<mailto:ith...@uoguelph.ca> Evidently this is a known bug: https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt97709<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fbst.cloudapps.cisco.com%2Fbugsearch%2Fbug%2FCSCvt97709=04%7C01%7C%7C3df5a410041c4c9f3aa308d98352ff54%7C84df9e7fe9f640afb435%7C1%7C0%7C637685214764520958%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=yMWZmOQ6hzS19Y4Xszh%2FvkymHnSUjYxxvHXQbFDngQ4%3D=0> The “solution” is to do the upgrade, then do an export and a full redeployment with a new OVA template. Since there are trust, certs, etc…involved in this, it is a very involved procedure. Ughhh… Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.ox.com%2F=04%7C01%7C%7C3df5a410041c4c9f3aa308d98352ff54%7C84df9e7fe9f640afb435%7C1%7C0%7C637685214764530920%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=bKsKDAT%2BmUil7SZAEa47SIzhDooeI1%2BYUWX1OOUP7A0%3D=0> ... From: Tim Smith mailto:tim.sm...@enject.com.au>> Sent: Wednesday, September 29, 2021 8:05 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: James Buchanan mailto:james.buchan...@gmail.com>>; Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM I've had these alerts pop up on fresh install during a PCD migration (new VM's) (it was 12.5 from memory) I spent a bit of time back and forth with TAC and checking with other TAC engineers etc. I think the outcome was basically that it was the standard setup of the Active Partition, and it isn't really an issue, except that you will get those alerts (and you can adjust the thresholds) But it's only a partition inside, and basically you have other space available. What does your show diskusage look like? On Wed, 29 Sept 2021 at 21:56, Matthew Huff mailto:mh...@ox.com>> wrote: I deleted the old firmware files and we don’t use any MOH sources other than the built in one. The coppfile is for cleaning up the common partition, but I ran it anyway. The active parttion still shows 98% used. I’m afraid TAC is correct, the older ova file that we used for 11.5 created too
Re: [cisco-voip] Low space on partitions in CUCM
Evidently this is a known bug: https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt97709 The “solution” is to do the upgrade, then do an export and a full redeployment with a new OVA template. Since there are trust, certs, etc…involved in this, it is a very involved procedure. Ughhh… Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Tim Smith Sent: Wednesday, September 29, 2021 8:05 AM To: Matthew Huff Cc: James Buchanan ; Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM I've had these alerts pop up on fresh install during a PCD migration (new VM's) (it was 12.5 from memory) I spent a bit of time back and forth with TAC and checking with other TAC engineers etc. I think the outcome was basically that it was the standard setup of the Active Partition, and it isn't really an issue, except that you will get those alerts (and you can adjust the thresholds) But it's only a partition inside, and basically you have other space available. What does your show diskusage look like? On Wed, 29 Sept 2021 at 21:56, Matthew Huff mailto:mh...@ox.com>> wrote: I deleted the old firmware files and we don’t use any MOH sources other than the built in one. The copp file is for cleaning up the common partition, but I ran it anyway. The active parttion still shows 98% used. I’m afraid TAC is correct, the older ova file that we used for 11.5 created too small of an active partition for 14, and the only solution is to do a complete rebuild. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 7:37 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Oh right... 1. Delete old firmware files. 2. Delete unused MOH sources. 3. Use the free space .cop file. On Wed, Sep 29, 2021 at 12:32 PM Matthew Huff mailto:mh...@ox.com>> wrote: The alert that is being triggered isn’t the LogPartition space, it’s the active partition space, which I believe is a separate area. Any suggestion on how to cleanup space on the active parttion? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:39 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, I totally agree but I think you'll be OK. I've done it when at a similar level and it's helped. Another thing you might do is to download all the log files and set them to delete when you download them. That might get you at a safer level. Kind Regards, James On Wed, Sep 29, 2021 at 11:36 AM Matthew Huff mailto:mh...@ox.com>> wrote: If it was like 92%, fixing the log threshold would make me feel less nervous. At 98%, I’m more concerned. Maybe I shouldn’t be? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:33 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, Have you tried adjusting the log file thresholds in RTMT? See https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200581-Procedure-to-Adjust-WaterMark-in-RTMT-of.html for more details. Thanks, James On Wed, Sep 29, 2021 at 11:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that
Re: [cisco-voip] Low space on partitions in CUCM
It’s 98% on active partion, 94% on inactive one, on both publisher and subscriber. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Tim Smith Sent: Wednesday, September 29, 2021 8:05 AM To: Matthew Huff Cc: James Buchanan ; Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM I've had these alerts pop up on fresh install during a PCD migration (new VM's) (it was 12.5 from memory) I spent a bit of time back and forth with TAC and checking with other TAC engineers etc. I think the outcome was basically that it was the standard setup of the Active Partition, and it isn't really an issue, except that you will get those alerts (and you can adjust the thresholds) But it's only a partition inside, and basically you have other space available. What does your show diskusage look like? On Wed, 29 Sept 2021 at 21:56, Matthew Huff mailto:mh...@ox.com>> wrote: I deleted the old firmware files and we don’t use any MOH sources other than the built in one. The copp file is for cleaning up the common partition, but I ran it anyway. The active parttion still shows 98% used. I’m afraid TAC is correct, the older ova file that we used for 11.5 created too small of an active partition for 14, and the only solution is to do a complete rebuild. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 7:37 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Oh right... 1. Delete old firmware files. 2. Delete unused MOH sources. 3. Use the free space .cop file. On Wed, Sep 29, 2021 at 12:32 PM Matthew Huff mailto:mh...@ox.com>> wrote: The alert that is being triggered isn’t the LogPartition space, it’s the active partition space, which I believe is a separate area. Any suggestion on how to cleanup space on the active parttion? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:39 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, I totally agree but I think you'll be OK. I've done it when at a similar level and it's helped. Another thing you might do is to download all the log files and set them to delete when you download them. That might get you at a safer level. Kind Regards, James On Wed, Sep 29, 2021 at 11:36 AM Matthew Huff mailto:mh...@ox.com>> wrote: If it was like 92%, fixing the log threshold would make me feel less nervous. At 98%, I’m more concerned. Maybe I shouldn’t be? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:33 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, Have you tried adjusting the log file thresholds in RTMT? See https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200581-Procedure-to-Adjust-WaterMark-in-RTMT-of.html for more details. Thanks, James On Wed, Sep 29, 2021 at 11:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Te
Re: [cisco-voip] Low space on partitions in CUCM
I deleted the old firmware files and we don’t use any MOH sources other than the built in one. The copp file is for cleaning up the common partition, but I ran it anyway. The active parttion still shows 98% used. I’m afraid TAC is correct, the older ova file that we used for 11.5 created too small of an active partition for 14, and the only solution is to do a complete rebuild. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan Sent: Wednesday, September 29, 2021 7:37 AM To: Matthew Huff Cc: Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM Oh right... 1. Delete old firmware files. 2. Delete unused MOH sources. 3. Use the free space .cop file. On Wed, Sep 29, 2021 at 12:32 PM Matthew Huff mailto:mh...@ox.com>> wrote: The alert that is being triggered isn’t the LogPartition space, it’s the active partition space, which I believe is a separate area. Any suggestion on how to cleanup space on the active parttion? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:39 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, I totally agree but I think you'll be OK. I've done it when at a similar level and it's helped. Another thing you might do is to download all the log files and set them to delete when you download them. That might get you at a safer level. Kind Regards, James On Wed, Sep 29, 2021 at 11:36 AM Matthew Huff mailto:mh...@ox.com>> wrote: If it was like 92%, fixing the log threshold would make me feel less nervous. At 98%, I’m more concerned. Maybe I shouldn’t be? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:33 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, Have you tried adjusting the log file thresholds in RTMT? See https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200581-Procedure-to-Adjust-WaterMark-in-RTMT-of.html for more details. Thanks, James On Wed, Sep 29, 2021 at 11:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Low space on partitions in CUCM
The alert that is being triggered isn’t the LogPartition space, it’s the active partition space, which I believe is a separate area. Any suggestion on how to cleanup space on the active parttion? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan Sent: Wednesday, September 29, 2021 6:39 AM To: Matthew Huff Cc: Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, I totally agree but I think you'll be OK. I've done it when at a similar level and it's helped. Another thing you might do is to download all the log files and set them to delete when you download them. That might get you at a safer level. Kind Regards, James On Wed, Sep 29, 2021 at 11:36 AM Matthew Huff mailto:mh...@ox.com>> wrote: If it was like 92%, fixing the log threshold would make me feel less nervous. At 98%, I’m more concerned. Maybe I shouldn’t be? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan mailto:james.buchan...@gmail.com>> Sent: Wednesday, September 29, 2021 6:33 AM To: Matthew Huff mailto:mh...@ox.com>> Cc: Cisco VoIP Group mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, Have you tried adjusting the log file thresholds in RTMT? See https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200581-Procedure-to-Adjust-WaterMark-in-RTMT-of.html for more details. Thanks, James On Wed, Sep 29, 2021 at 11:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Low space on partitions in CUCM
If it was like 92%, fixing the log threshold would make me feel less nervous. At 98%, I’m more concerned. Maybe I shouldn’t be? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: James Buchanan Sent: Wednesday, September 29, 2021 6:33 AM To: Matthew Huff Cc: Cisco VoIP Group Subject: Re: [cisco-voip] Low space on partitions in CUCM Hi Matthew, Have you tried adjusting the log file thresholds in RTMT? See https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200581-Procedure-to-Adjust-WaterMark-in-RTMT-of.html for more details. Thanks, James On Wed, Sep 29, 2021 at 11:28 AM Matthew Huff mailto:mh...@ox.com>> wrote: After upgrading from 11.5 to 14, I’m at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I’m loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Low space on partitions in CUCM
After upgrading from 11.5 to 14, I'm at 98% on publisher/subscriber on the active partition. Cisco TACs only suggestion is to redeploy from scratch from 14 ova. Given the issues with certs, trusts, licensing, etc, I'm loathe to do that. Anyone have any suggestions? Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] [External] Non-jabber based Cisco softphones for iPhone and Android
Cool, Thanks. That looks like at the very least a quick and cheap solution. Matthew Huff | Director of Technical Operations | OTA Management LLC Office: 914-460-4039 mh...@ox.com<mailto:mh...@ox.com> | www.ox.com<http://www.ox.com> ... From: Josh Nordquist Sent: Monday, February 1, 2021 11:27 AM To: Matthew Huff Cc: Johnson, Tim ; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] [External] Non-jabber based Cisco softphones for iPhone and Android We don't have an IM server so I'm assuming it isn't needed and you only need an expressway for remote access. We used the mobile clients internally on wifi networks before COVID and added that later. On Mon, Feb 1, 2021 at 10:12 AM Matthew Huff mailto:mh...@ox.com>> wrote: Thanks, another issue with the jabber client is what you mention. The lack of additional lines. We need something that works more like a softphone and less like an IM client. With the jabber phone only config, do you still need an IM server, or just the MRA? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] [External] Non-jabber based Cisco softphones for iPhone and Android
Thanks, another issue with the jabber client is what you mention. The lack of additional lines. We need something that works more like a softphone and less like an IM client. With the jabber phone only config, do you still need an IM server, or just the MRA? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] [External] Non-jabber based Cisco softphones for iPhone and Android
Last I tried it still required an IM presence server would entail a large infrastructure cost because even if we disable IM, FINRA would require us to still have archiving done in case it got turned back on. ... -Original Message- From: Johnson, Tim Sent: Monday, February 1, 2021 11:00 AM To: Matthew Huff ; cisco-voip@puck.nether.net Subject: RE: [External] [cisco-voip] Non-jabber based Cisco softphones for iPhone and Android Pretty sure you can run Jabber in phone-only mode these days. Not sure on the limitations of it though. -Original Message- From: cisco-voip On Behalf Of Matthew Huff Sent: Monday, February 1, 2021 10:56 AM To: cisco-voip@puck.nether.net Subject: [External] [cisco-voip] Non-jabber based Cisco softphones for iPhone and Android We are looking for a solution for softphones that are compatible with expressway for both iPhones and android. Jabber is not an option due to Finra & SEC compliance (no instant messaging). Specifically we are looking for a voice app only with no collaboration/presence features. Any suggestions? ___ 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] Non-jabber based Cisco softphones for iPhone and Android
We are looking for a solution for softphones that are compatible with expressway for both iPhones and android. Jabber is not an option due to Finra & SEC compliance (no instant messaging). Specifically we are looking for a voice app only with no collaboration/presence features. Any suggestions? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Removing last IM & Presence server
Just found docs on removing nodes from Presence cluster, not removing the last one. I removed all the SRV records. Doing a full cluster reboot is probably a good idea to do this weekend. Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] Sent: Tuesday, January 8, 2019 10:23 AM To: Matthew Huff Cc: cisco-voip (cisco-voip@puck.nether.net) Subject: Re: [cisco-voip] Removing last IM & Presence server There might be UC Services created for this server in CUCM (which would then be apart of one or more Service Profiles). If you're using the _cup-login SRV DNS record, remove it from there. If you're using multi-server certificates, then you might also need/want to re-do your certs, to remove it's name from those, but I don't really think that's a critical task, and could likely just wait until your certs are up for renewal. You might also want to look at a cluster reboot, just to clear things up, and kick off some of those admin scripts which seem to run on reboot. I should probably read the docs before giving advice, but those things just popped into my head. Have you read the documentation on this procedure yet? Is there even any? Can you link what you found? On Tue, Jan 8, 2019 at 8:44 AM Matthew Huff mailto:mh...@ox.com>> wrote: Other than un-assigning the users, deleting the prescense redundancy group and deleting the server from the Unified CM administrator, is there anything I need to do to cleanup if we are no longer using IM & P? ---- Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Removing last IM & Presence server
Other than un-assigning the users, deleting the prescense redundancy group and deleting the server from the Unified CM administrator, is there anything I need to do to cleanup if we are no longer using IM & P? ---- Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Jabber 11.8 issue
Another item to check: o) If you are using CUCM for service discovery (have a SRV record for _cisco-uds._tcp.), then in order to do discovery the user account in CUCM has to be a member of the "Standard CCM End User" access group in order to be able to access the XML configuration file to discover the jabber server. ---- Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 -Original Message- From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Pawlowski, Adam Sent: Friday, October 20, 2017 1:03 PM To: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber 11.8 issue Norm, I went through some materials and had documented the causes for that message (which show up in a PRT) as best I can. "Cannot communicate with server" in my notes always boils down to the workstation not actually being able to reach the server for whatever reason. Either whatever SRV record it sees is pointing it somewhere it cannot go, or the other option is the " ServiceDiscoveryNoNetworkConnectivity " error wherein Jabber believes that it has no network connectivity. I haven't seen that one on sign in but have seen it on resume on Mac before. Jabber also generates this message if you set the minimum version in the IM and Presence server to something like "11.8.4" but the Mac version never made it past 11.8.1 - if the version isn't allowed it says this. I don't believe it says this if you don't check the "Home Cluster" box somewhere in the end user for the user to have a home cluster but it will throw an error for that as well. Regards, Adam Pawlowski Network and Classroom Services University at Buffalo 716.645.8489 ___ 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] Skipping * on inbound external calls to Unity
That’s exactly what I have. However, when you dial that DID, you get a prompt asking you to enter the phone number you want to leave a message for instead of asking you to sign-in. However, if you type “*”, it will let you sign in. From google, this appears to be normal issue and the recommendation is to train users to know that. Our users aren’t that trainable however. Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 From: Charles Goldsmith [mailto:wo...@justfamily.org] Sent: Tuesday, October 3, 2017 12:41 PM To: Matthew Huff <mh...@ox.com> Cc: cisco-voip voyp list <cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Skipping * on inbound external calls to Unity IIRC, you should be able to setup a direct routing rule with the DID and have it send to Conversation > Attempt Sign-In. On CUCM, just setup a CTI-RP to send the number over to CUC. On Tue, Oct 3, 2017 at 10:52 AM, Matthew Huff <mh...@ox.com<mailto:mh...@ox.com>> wrote: If we have an externally available number to access Unity voicemail, is there anyway of changing it so that it doesn't prompt asking for phone number to dial? You can hit * to enter your extension and passcode, but only if you remember to hit *. Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Skipping * on inbound external calls to Unity
If we have an externally available number to access Unity voicemail, is there anyway of changing it so that it doesn't prompt asking for phone number to dial? You can hit * to enter your extension and passcode, but only if you remember to hit *. Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Naming conventions for Expressway-C / E
What about in a cluster? Would it be a good idea to name it: ny-expc-01.domain.com<http://ny-expc-01.domain.com> ny-expc-02.domain.com<http://ny-expc-02.domain.com> ny-expe-01.domain.com<http://ny-expe-01.domain.com> ny-expe-02.domain.com<http://ny-expe-02.domain.com> What about public/private names? On Sep 4, 2017, at 7:58 AM, Bill Talley <btal...@gmail.com<mailto:btal...@gmail.com>> wrote: Similar to Ryan, I usually try to include "expe" or "expc" as well when the longer name isn't allowed, depending on customers naming standards. Sent from a mobile device with very tiny touchscreen input keys. Please excude my typtos. On Sep 4, 2017, at 7:05 AM, Ryan Huff <ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote: I usually keep the names, "edge" and "control" incorporated in them somehow. If I'm trying to save on typing and it's a single pair I might do, "expcontrol" and "expedge". With clusters, I'll add more to the name that denote its purpose or location. Thanks, Ryan On Sep 4, 2017, at 6:17 AM, Matthew Huff <mh...@ox.com<mailto:mh...@ox.com>> wrote: I’m about to deploy our first expressway cluster, what names do people use form expressway DNS names? ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Naming conventions for Expressway-C / E
I'm about to deploy our first expressway cluster, what names do people use form expressway DNS names? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Multiple SIP MWI sources with CME
We are running CME 11.6 and are in the process of migrating from CUE to Unity Connection. Everything is working, but during the testing, is there any way I can configure CME to have two sources of MWI? The Unity Connection is sending unsolicited SIP messages for the MWI, but the CME is ignoring it since it already has CUE setup as the SIP MWI source and you can't add two sources. Any possible solution? It will make the migration easier if we can have both. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server
Yep, cisco TAC just told me that also. Evidently the error is pretty generic and related to issues on the pub. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: Friday, August 25, 2017 10:12 AM To: Anthony Holloway <avholloway+cisco-v...@gmail.com>; Matthew Huff <mh...@ox.com>; cisco-voip list <cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server The Unified Communications IM and Presence ISO is bootable I believe (maybe that changed?); although it's sister ISO, Unified Communications Manager is not. From: cisco-voip <cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> on behalf of Anthony Holloway <avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> Sent: Friday, August 25, 2017 10:08 AM To: Matthew Huff; cisco-voip list Subject: Re: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server I don't have an answer for you, but you reminded me that UCCX 11.6 is setting the new standard for ISOs. The UCCX 11.6 ISO on CCO for upgrading, is also bootable, and can be used for fresh installs as well. Get on board IM, so people don't have to create their own bootable media. Single ISO for Upgrades and Fresh-Install In UCCX 11.6, there is only 1 ISO released that is posted on Cisco.com and this ISO can be used for either an upgrade or a fresh install. The ISO follows the regular naming convention of UCSInstall_UCCX _11.6.X-XX.sgn.iso This ISO is provided with both boot options, so serves as a bootable image as well. Source: https://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express/211582-Tech-Note-on-UCCX-11-6-Pre-Release-Commu.html On Thu, Aug 24, 2017 at 1:13 PM Matthew Huff <mh...@ox.com<mailto:mh...@ox.com>> wrote: I've tried repeatedly to install our second IM server, and it always fails with a corrupt file during post-install. I re-downloaded the ISO and recreated the bootable ISO, so I don't believe it's a ISO issue. Anyone see this? ___ cisco-voip mailing list cisco-voip@puck.nether.net<mailto: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] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server
Thanks Ben! That was it. Didn't occur to me that there might be a problem in the ova template. Perhaps the secondary has to replicate during installation and there wasn't enough disk space? The BE6K comes with version 1.2 of the ova template, but there is a 1.4 version on CCO. The installation hasn't finished yet, but it's much further along than it been before. From: Matthew Huff Sent: Friday, August 25, 2017 5:19 AM To: 'Ben Amick' <bam...@humanarc.com>; cisco-voip list <cisco-voip@puck.nether.net> Subject: RE: Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server Yes, It's a BE6k. Haven't thought to re-download the template, worth a try. From: Ben Amick [mailto:bam...@humanarc.com] Sent: Thursday, August 24, 2017 3:26 PM To: Matthew Huff <mh...@ox.com<mailto:mh...@ox.com>>; cisco-voip list <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: RE: Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server Just curious, is your first server from the BE6/7000 Preconfigures? I remember having a few major issues in the past adding servers to a cluster from one of the preconfigured templates on 10.5 and 11.0, wondering if it might be that same issue. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Matthew Huff Sent: Thursday, August 24, 2017 2:13 PM To: cisco-voip list <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server I've tried repeatedly to install our second IM server, and it always fails with a corrupt file during post-install. I re-downloaded the ISO and recreated the bootable ISO, so I don't believe it's a ISO issue. Anyone see this? 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
Re: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server
Yes, It's a BE6k. Haven't thought to re-download the template, worth a try. From: Ben Amick [mailto:bam...@humanarc.com] Sent: Thursday, August 24, 2017 3:26 PM To: Matthew Huff <mh...@ox.com>; cisco-voip list <cisco-voip@puck.nether.net> Subject: RE: Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server Just curious, is your first server from the BE6/7000 Preconfigures? I remember having a few major issues in the past adding servers to a cluster from one of the preconfigured templates on 10.5 and 11.0, wondering if it might be that same issue. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Matthew Huff Sent: Thursday, August 24, 2017 2:13 PM To: cisco-voip list <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: [cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server I've tried repeatedly to install our second IM server, and it always fails with a corrupt file during post-install. I re-downloaded the ISO and recreated the bootable ISO, so I don't believe it's a ISO issue. Anyone see this? 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
[cisco-voip] Cisco IM & Presence 11 5.1 SU2 corrupt install with 2nd server
I've tried repeatedly to install our second IM server, and it always fails with a corrupt file during post-install. I re-downloaded the ISO and recreated the bootable ISO, so I don't believe it's a ISO issue. Anyone see this? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CCX and NTP
If you are going to have local NTP servers, you will need 3 for a quorum. 4 for redundancy. Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan Huff Sent: Monday, May 8, 2017 10:42 AM To: Ben Amick <bam...@humanarc.com>; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CCX and NTP Hey Ben, This has less to do with UCCX and more to do with the lack of time precision in Windows SNTP / NTP compared to the needs of UCOS. Yes, I know, die hard Redmond boys and girls will say that ever since Win2003 that isn't the case but sorry, it is. (read: https://technet.microsoft.com/en-us/library/cc773013%28v=ws.10%29.aspx). In fact, on page 84 (PDF page) of the latest SRND (http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11.pdf) Windows NTP is a flat nono for Cisco UC period. IOS routers are great but here is how I like to roll (excuse the crudeness of the drawing, put it together this morning before coffee). I typically use Linux servers for NTP because they are so darn useful for other utilitarian things . can't count how many times just having a Linux box on the network somewhere helped me out. [cid:image001.png@01D2C7E8.B89D1C90] = Ryan = From: cisco-voip <cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> on behalf of Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>> Sent: Monday, May 8, 2017 10:12 AM To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> Subject: [cisco-voip] CCX and NTP What do you guys use for NTP on your CCX hosts? I've been informed by TAC that "CCX does not support Windows based NTP" so I was thinking about just pointing NTP towards my CCM hosts - is that a valid scenario? I figure that since CCM is pretty much authoritative on everything for CCX as it is it wouldn't be a problem? 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
Re: [cisco-voip] Curiosity: New IP Communicator
Okay, so, I’ve been until recently a CME admin, now I’m doing CUCM, so…. What’s MRA? Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade Sent: Tuesday, February 28, 2017 2:11 PM To: Ben Amick <bam...@humanarc.com> Cc: Matthew Huff <mh...@ox.com>; Cisco VOIP <cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Curiosity: New IP Communicator Yes, they do support SSL VPN and MRA although MRA has a smaller featureset. On Tue, Feb 28, 2017 at 1:12 PM, Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>> wrote: I'm fairly certain both the 7800 and 8800 phones support Cisco VPN, and I think they also support expressway over MRA. Could be wrong, but I vaguely recall that. Ben Amick Telecom Analyst On Feb 28, 2017, at 1:09 PM, Matthew Huff <mh...@ox.com<mailto:mh...@ox.com>> wrote: We use SCCP SecureVPN phones now for remote access for SOHO users. We would like to use newer SIP phones, but so far, I haven’t seen a solution for the 88xx phones. We also need multiple lines, shared lines, hunt groups, etc… As far as I know ExpressWay is only for Jabber clients, does anyone know of a solution for Cisco 88xx phones secure remote access? Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039<tel:(914)%20460-4039> aim: matthewbhuff| Fax: 914-694-5669<tel:(914)%20694-5669> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ben Amick Sent: Tuesday, February 28, 2017 12:54 PM To: Charles Goldsmith <wo...@justfamily.org<mailto:wo...@justfamily.org>>; Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>> Cc: Cisco VOIP <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Curiosity: New IP Communicator I’ll raise my hand as one of those contact centers. Right now it’s because of the two reasons I mentioned in a parallel chain: a need for multiple lines and the BIB for QM recording. The former is on the roadmap for jabber (with no ETA), but the latter as far as I’m aware still isn’t a part of Jabber. Ben Amick Telecom Analyst From: Charles Goldsmith [mailto:wo...@justfamily.org] Sent: Tuesday, February 28, 2017 12:49 PM To: Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>> Cc: Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>>; Cisco VOIP <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Curiosity: New IP Communicator There have been a few updates to CIPC in the last couple of years, but I haven't read the release notes as to what, so someone has been spending a bit of time on the code. I've seen a lot of contact centers use CIPC, I don't see it going away soon. On Tue, Feb 28, 2017 at 11:21 AM, Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>> wrote: My understanding is that the answer is “No,” to updated IP communicator. I am also not sure of the true commitment to multi-line Jabber support. I have yet to see any dates around multi-line Jabber at this point, which my experience with Cisco say’s don’t count on it or expect it. Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814<tel:(314)%20212-1814> <http://cp.mcafee.com/d/avndy0w72gQrhouupKMyYYrKrhhpvpj73AjhOrhhpvpj7ffICQkmnTDNPPXxJ55MQsCzCZXETdAdlBoG2yrqKMSdKndASRtxIrsKrIOYgv8ffZvDTPhOYMCZRXBQSnSud7b3Xb9EVoWyaqRQRrFIYG7DR8OJMddECSjt-jLuZXTLuVKVIDeqR4IOpykV_7BY4GDDWg6fxjV6P5iFngk8JQCXGIzOT00jrdQm4jobZ8Qg6BKQGmGncRAIqnjh09dNBmAFBAzh03fc6y0d2NEw4GGGJMBrpRyro76SZSgWJOE> <tel:+13142121814> "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<http://cp.mcafee.com/d/1jWVIi4zqb3PPdS4nDztPqabbXaoUsyqejqabbXaoVVZASyyO-Y-euvsdEEK6zAQsTLt6VIxGIH5gkjrlS6NJOVICSHIdzrBPtCny3V1V_HY--qenC4TKLsKCO-PNEVovppd7b7khjmKCHtdDBgY-F6lK1FJcSOrLOtXTLuZXTdTdAVPmEBC634-05UvO-a_YYg-49kUDYBjpyFkHEa4mWjtRmhVrw09JCWb29I5-Aq83iTqlblbCqOmdbFEw4CUOHikOOhEw1DC3h06xoQg2lllmUiJIWNdI3zoAihZcj
Re: [cisco-voip] Curiosity: New IP Communicator
We use SCCP SecureVPN phones now for remote access for SOHO users. We would like to use newer SIP phones, but so far, I haven’t seen a solution for the 88xx phones. We also need multiple lines, shared lines, hunt groups, etc… As far as I know ExpressWay is only for Jabber clients, does anyone know of a solution for Cisco 88xx phones secure remote access? Matthew Huff | 1 Manhattanville Rd Director of Operations | Purchase, NY 10577 OTA Management LLC | Phone: 914-460-4039 aim: matthewbhuff| Fax: 914-694-5669 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ben Amick Sent: Tuesday, February 28, 2017 12:54 PM To: Charles Goldsmith <wo...@justfamily.org>; Heim, Dennis <dennis.h...@wwt.com> Cc: Cisco VOIP <cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Curiosity: New IP Communicator I’ll raise my hand as one of those contact centers. Right now it’s because of the two reasons I mentioned in a parallel chain: a need for multiple lines and the BIB for QM recording. The former is on the roadmap for jabber (with no ETA), but the latter as far as I’m aware still isn’t a part of Jabber. Ben Amick Telecom Analyst From: Charles Goldsmith [mailto:wo...@justfamily.org] Sent: Tuesday, February 28, 2017 12:49 PM To: Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>> Cc: Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>>; Cisco VOIP <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: Re: [cisco-voip] Curiosity: New IP Communicator There have been a few updates to CIPC in the last couple of years, but I haven't read the release notes as to what, so someone has been spending a bit of time on the code. I've seen a lot of contact centers use CIPC, I don't see it going away soon. On Tue, Feb 28, 2017 at 11:21 AM, Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>> wrote: My understanding is that the answer is “No,” to updated IP communicator. I am also not sure of the true commitment to multi-line Jabber support. I have yet to see any dates around multi-line Jabber at this point, which my experience with Cisco say’s don’t count on it or expect it. Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814<tel:(314)%20212-1814> [cid:image001.png@01D10DD2.7FC81F90]<http://cp.mcafee.com/d/avndy0w72gQrhouupKMyYYrKrhhpvpj73AjhOrhhpvpj7ffICQkmnTDNPPXxJ55MQsCzCZXETdAdlBoG2yrqKMSdKndASRtxIrsKrIOYgv8ffZvDTPhOYMCZRXBQSnSud7b3Xb9EVoWyaqRQRrFIYG7DR8OJMddECSjt-jLuZXTLuVKVIDeqR4IOpykV_7BY4GDDWg6fxjV6P5iFngk8JQCXGIzOT00jrdQm4jobZ8Qg6BKQGmGncRAIqnjh09dNBmAFBAzh03fc6y0d2NEw4GGGJMBrpRyro76SZSgWJOE> [cid:image002.png@01D10DD2.7FC81F90][cid:image003.png@01D10DD2.7FC81F90]<tel:+13142121814>[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<http://cp.mcafee.com/d/1jWVIi4zqb3PPdS4nDztPqabbXaoUsyqejqabbXaoVVZASyyO-Y-euvsdEEK6zAQsTLt6VIxGIH5gkjrlS6NJOVICSHIdzrBPtCny3V1V_HY--qenC4TKLsKCO-PNEVovppd7b7khjmKCHtdDBgY-F6lK1FJcSOrLOtXTLuZXTdTdAVPmEBC634-05UvO-a_YYg-49kUDYBjpyFkHEa4mWjtRmhVrw09JCWb29I5-Aq83iTqlblbCqOmdbFEw4CUOHikOOhEw1DC3h06xoQg2lllmUiJIWNdI3zoAihZcjt> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>] On Behalf Of Ben Amick Sent: Tuesday, February 28, 2017 11:45 AM To: Cisco VOIP <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>> Subject: [cisco-voip] Curiosity: New IP Communicator I’m assuming no on this question, based on the fact that I haven’t heard even an inkling, and I know they’re working on multi-line support for Jabber, but I was wondering: Has anyone heard anything about them possibly coming up with a new IP communicator that is based on the 8800 series code? I think that’d be great, and even better if they made it based on 8845 code, so it could use video as well, short of getting everyone to use jabber. 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 rec
