Re: [cisco-voip] Unusual UCOS installation condition - sanity check please
Have you tried changing the mac address of your VM? Sometimes it just happen to generate the same Mac Address as other VM in the network. Regards, Ki Wi On Sun, Mar 20, 2016 at 11:55 PM, Ryan Huffwrote: > Encountering the *duplicate ip on eth0* installation error for UCOS. The > IP is not actually duplicated on the segment though. > > > My experience with this error is that it is usually right; in the sense > that * something* is not compatible with UCOS in the network. What I'm > not certain of, is what method the installer uses to determine the > duplicate condition. I believe the installer is using an ARP ping to > determine if there is a dup condition. The error comes on a subscriber, > right after the validation to first node check. > > > I believe the network may be using Proxy ARP (not completely sure); which > to my knowledge, won't work with UCOS. I don't have visibility/access into > the network but I do need to be able to articulate *why* the > installer detects a dup IP (when there really isn't a dup IP). I have > quadruple verified I am using the right first node / security password. > This is happening for multiple subscriber nodes too (same segment), so I'm > confident it is network related, not fat-finger related. > > > Here is the Scenario: > >- CUCM 10.5.2 cluster over WAN (not sure what the WAN link is) >- 2 sites; SITE_A and SITE_B (not sure what infrastructure gear is at >each site) > - SITE_A: pub > - SITE_B: sub > > In SITE_A, the pub install without issue. In SITE_B, right after the > sub installer does the validation to first node check, it pops the *duplicate > ip* error. I have a linux server installed on the SITE_A and SITE_B > segment. > > > From the SITE_A linux server, I do an ARPing to the SITE_B IP addresses > and I get the MAC address for the SITE_A WAN router (Cisco OID). From the > SITE_B linux server, I do an ARPing to the SITE_A IP addresses and I get > the MAC address for the SITE_B WAN router (Cisco OID). > > > Anyone have any thoughts into this? > > Thanks, > > Ryan > > ___ > 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] RTMT alert
Hi Anthony, thanks for your confirmation! Cheers, Ki Wi On Mon, Mar 14, 2016 at 10:09 PM, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Ki Wi, > > RTMT does not need to be "on" or running in order for alerts to be > triggered and emails to be sent out; custom or otherwise. > > On Mon, Mar 14, 2016 at 3:36 AM, 秀王 <kiwi.vo...@gmail.com> wrote: > >> Does RTMT application needs to be on for >> >> 1) pre-defined alert? >> 2) user defined alert? >> >> >> For example, this case. I suppose this is pre-defined alert so automated >> e-mail is possible? >> >> >> Configure the managed file transfer service parameters to define the >> threshold at which an RTMT alarm is generated for the external file server >> disk space. >> >>1. Log in to the node's Cisco Unified CM IM and Presence >>Administration user interface. >>2. Choose System > Service Parameters. >>3. Choose the Cisco XCP File Transfer Manager service for the node. >>4. Enter the required percentage values for the External File Server >>Available Space Lower Threshold and External File Server Available >>Space Upper Threshold service parameters. >>5. Choose Save. >> >> Cheers, >> Ki Wi >> >> >> >> ___ >> 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] Migration strategy
hi guys, thanks! gonna try them in the lab first to figure out if i can put the device profile in the production partition. On Wed, Mar 18, 2015 at 8:58 PM, Rob Dawson rdaw...@force3.com wrote: This is the way I handle scenarios like this as well – create a temporary partition for the DNs, etc. and use BAT to bulk edit them when it is time to move to production. Having the profiles exposed shouldn’t cause any issues as long as the DNs are hidden. Rob *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf Of *Ryan Huff *Sent:* Wednesday, March 18, 2015 8:06 AM *To:* kiwi.vo...@gmail.com *Cc:* cisco-voip@puck.nether.net *Subject:* Re: [cisco-voip] Migration strategy Any elements that you are going to pre-stage that are a part of the dial plan (translations, route patterns, DNs, transformations ...etc) wilk all need to be isolated from your currently migrated phones. So the DNs on your device profiles would likely need to be in an isolated partition, but the device profile itself probably doesn't need to be in an isolated device pool. If you do this, remember to use a CLLI or some unique descriptor in the description field of all your pre-staged elements, it will make it very easy for BAT to find things later on. Thanks, Ryan Original Message From: 秀王 kiwi.vo...@gmail.com Sent: Tuesday, March 17, 2015 11:32 PM To: Ryan Huff ryanh...@outlook.com Subject: Re: [cisco-voip] Migration strategy CC: cisco-voip@puck.nether.net Hi Ryan, let's say my phones are on a temp partition not reachable by other CSS. My UDP (user device profile) are on a valid partition shared by others ( Ie. P_Internal) but they are not logged in anywhere. Will this confused the CUCM? Or i shall place the UDP on temp partition as well. If so, can BAT assist me in migrating from TEMP partition to actual partition (P_Internal)? Cheers, Ki Wi On Wed, Mar 18, 2015 at 9:26 AM, Ryan Huff ryanh...@outlook.com wrote: I'm not, sure I completely understand your questions but I'll attempt to answer based on my understanding. Yes, you can pre-config devices and users in CCM prior to migration. If they are Cisco IP phones, you'll need the MAC address and model of the phone at a minimum. If they are non Cisco IP phones, you'll need to pre configure 3rd party sip devices (which is a different license requirement than a Cisco IP phone). Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. partition that the already migrated phones cannot access. As you migrate, change that partition using BAT, to the correct partition for the portion of phones you migrated. Thanks, Ryan Original Message From: 秀王 kiwi.vo...@gmail.com Sent: Tuesday, March 17, 2015 09:15 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Migration strategy Currently the client have avaya and cisco linked together using SIP. Cisco UCM cluster have users in the production environment. I'm are going to cutover more sites from avaya to cisco. Is it possible to preconfigure the users, extension number (let's say 87XXX range), phones and the user device profiles in advance? I'm thinking that if I preconfigure those information, the cucm will think that those extension number (87XXX) are local and unregistered. Is there a way to make CUCM thinks that in order to reach 87XXX range, it will still reach out to Avaya using the SIP trunk? Is there any setting in the route pattern can do that? I thinking that CUCM will always find a more exact match locally instead of through other source like translation pattern or route pattern. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Migration strategy
Hi Ryan, let's say my phones are on a temp partition not reachable by other CSS. My UDP (user device profile) are on a valid partition shared by others ( Ie. P_Internal) but they are not logged in anywhere. Will this confused the CUCM? Or i shall place the UDP on temp partition as well. If so, can BAT assist me in migrating from TEMP partition to actual partition (P_Internal)? Cheers, Ki Wi On Wed, Mar 18, 2015 at 9:26 AM, Ryan Huff ryanh...@outlook.com wrote: I'm not, sure I completely understand your questions but I'll attempt to answer based on my understanding. Yes, you can pre-config devices and users in CCM prior to migration. If they are Cisco IP phones, you'll need the MAC address and model of the phone at a minimum. If they are non Cisco IP phones, you'll need to pre configure 3rd party sip devices (which is a different license requirement than a Cisco IP phone). Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. partition that the already migrated phones cannot access. As you migrate, change that partition using BAT, to the correct partition for the portion of phones you migrated. Thanks, Ryan Original Message From: 秀王 kiwi.vo...@gmail.com Sent: Tuesday, March 17, 2015 09:15 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] Migration strategy Currently the client have avaya and cisco linked together using SIP. Cisco UCM cluster have users in the production environment. I'm are going to cutover more sites from avaya to cisco. Is it possible to preconfigure the users, extension number (let's say 87XXX range), phones and the user device profiles in advance? I'm thinking that if I preconfigure those information, the cucm will think that those extension number (87XXX) are local and unregistered. Is there a way to make CUCM thinks that in order to reach 87XXX range, it will still reach out to Avaya using the SIP trunk? Is there any setting in the route pattern can do that? I thinking that CUCM will always find a more exact match locally instead of through other source like translation pattern or route pattern. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UK PRI MGCP
Sorry to dig a very old thread. Is this method support today? Is it still limited to like maximum 5 field where we can put them into maintenance mode? On Thu, Dec 3, 2009 at 11:36 PM, VoiceNoob voicen...@gmail.com wrote: No you have done some tweaks somewhere to make a partial PRI work with MGCP. It does not work correctly without some changes. It may be supported by TAC but not supported by CUCM. J *From:* cisco-voip-boun...@puck.nether.net [mailto: cisco-voip-boun...@puck.nether.net] *On Behalf Of *Lewis, Chris *Sent:* Thursday, December 03, 2009 8:52 AM *To:* Charles Goldsmith; Joe Martini (joemar2) *Cc:* cisco-voip@puck.nether.net *Subject:* Re: [cisco-voip] UK PRI MGCP To let you know, I have been using 15 channels on an ISDN30 for over 2 years - in my UK site - with no additional tweaks on CCM or the VGW’s Plus it is supported by TAC – Spoken to Joe a few times I in the past J Chris *From:* cisco-voip-boun...@puck.nether.net [mailto: cisco-voip-boun...@puck.nether.net] *On Behalf Of *Charles Goldsmith *Sent:* 03 December 2009 14:36 *To:* Joe Martini (joemar2) *Cc:* cisco-voip@puck.nether.net *Subject:* Re: [cisco-voip] UK PRI MGCP *Note: *Cisco CallManager does not support the configuration or use of a fractional PRI when you use it with MGCP. If fractional PRI is necessary, you can use H.323 instead of MGCP. This can be found on http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00806fedbe.shtml#topic1 It's not supported and won't work without some tweaking. Charles On Thu, Dec 3, 2009 at 8:34 AM, Joe Martini (joemar2) joem...@cisco.com wrote: To do a fractional MGCP T1/E1/PRI you have to configured it as a full PRI on the gateway. Then on CUCM check the CallManager service parameter, Change B-Channel Maintenance Status. Joe *From:* cisco-voip-boun...@puck.nether.net [mailto: cisco-voip-boun...@puck.nether.net] *On Behalf Of *Dan Greenway *Sent:* Thursday, December 03, 2009 9:25 AM *To:* cisco-voip@puck.nether.net *Subject:* Re: [cisco-voip] UK PRI MGCP I didn't think you could do MGCP with fractional E1's thanks Dan On 3 Dec 2009, at 07:10, afatsum wrote: You are missing l3 backhaul binding to ccm-manager under your serial interface. Add isdn bind-l3 ccm-manager under serial 0/0/0:15 -Mus Martin Bufton wrote: I have a Cisco router connected to UK PRI E1 with 10 channels. I cannot get it to register. I get the following MCGP debug message. Can you see anything wrong in my config? 73: *Dec 2 17:55:59.251 GMT: //-1//MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid parameter (pkt 0x67654638 pkt-mgcp_parm_lines 0x) Thanks in advance Current configuration : 8969 bytes ! version 12.4 no service pad service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone service password-encryption service sequence-numbers ! hostname Cent-xxx-MGCP-GW1-RT ! boot-start-marker boot-end-marker ! card type e1 0 0 logging buffered 51200 warnings ! no aaa new-model clock timezone GMT 0 clock summer-time BST recurring last Sun Mar 2:00 last Sun Oct 2:00 network-clock-participate wic 0 dot11 syslog ! ! ip cef ! ! no ip bootp server no ip domain lookup ip domain name multilink bundle-name authenticated ! isdn switch-type primary-net5 voice-card 0 no dspfarm ! crypto pki trustpoint TP-self-signed-2117350501 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-2117350501 revocation-check none rsakeypair TP-self-signed-2117350501 quit ! ! archive log config hidekeys ! ! controller E1 0/0/0 pri-group timeslots 1-10,16 service mgcp ! controller E1 0/0/1 ! ip tcp synwait-time 10 ! ! ! ! interface Port-channel1 description EtherChannel to Cent-xxx-Core-Sw ip address 172.16.xx.100 255.255.255.0 ! interface GigabitEthernet0/0 description Member of Etherchannel conected to Cent-xxx-Core-Sw gig 2/0/19 no ip address duplex auto speed auto media-type rj45 channel-group 1 ! interface GigabitEthernet0/1 description Member of Etherchannel conected to Cent-xx-Core-Sw gig 1/0/19 no ip address duplex auto speed auto media-type rj45 channel-group 1 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 Port-channel1 172.16.xx.1 ! ! ip http server ip http access-class 23 ip http
[cisco-voip] Migration strategy
Currently the client have avaya and cisco linked together using SIP. Cisco UCM cluster have users in the production environment. I'm are going to cutover more sites from avaya to cisco. Is it possible to preconfigure the users, extension number (let's say 87XXX range), phones and the user device profiles in advance? I'm thinking that if I preconfigure those information, the cucm will think that those extension number (87XXX) are local and unregistered. Is there a way to make CUCM thinks that in order to reach 87XXX range, it will still reach out to Avaya using the SIP trunk? Is there any setting in the route pattern can do that? I thinking that CUCM will always find a more exact match locally instead of through other source like translation pattern or route pattern. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] SRST for Jabber Desktop
Hi Group members, Anyone tried SRST for Jabber? The client is using a cloud based IM (Webex messaging service) so SRST is supported on paper. Have yet to configure this previously. From what i have gathering so far, it seems like it is a pain in the arse to configure i suppose due to lack of information. Do i need to go for enhanced SRST in order to support Jabber Desktop for SRST? Seems like if the Jabber and the gateway is different subnet, i have to enable sip digest authentication as well. Lots of things to be configured just to achieve SRST for Jabber? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip