Try IE 11 or Firefox version 24 and above.  I've only ever used Firefox; that's 
what works for me.

> From: [email protected]
> Subject: cisco-voip Digest, Vol 134, Issue 2
> To: [email protected]
> Date: Tue, 2 Dec 2014 12:00:03 -0500
> 
> Send cisco-voip mailing list submissions to
>       [email protected]
> 
> To subscribe or unsubscribe via the World Wide Web, visit
>       https://puck.nether.net/mailman/listinfo/cisco-voip
> or, via email, send a message with subject or body 'help' to
>       [email protected]
> 
> You can reach the person managing the list at
>       [email protected]
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of cisco-voip digest..."
> 
> 
> Today's Topics:
> 
>    1. Expressway - 3rd Party Border Recommendation (Pawlowski, Adam)
>    2. Re: Expressway - 3rd Party Border Recommendation (Brian Meade)
>    3. Cisco Presence External Database Front End Application (Ryan Huff)
>    4. UC integration with MS Office and Lync/MOC
>       ([email protected])
>    5. H245Interface & Related Processes (Daniel Pagan)
>    6. Re: H245Interface & Related Processes (Ryan Ratliff (rratliff))
>    7. NFAS t1 (Charles Goldsmith)
>    8. Re: NFAS t1 (Charles Goldsmith)
>    9. Re: H245Interface & Related Processes (Brian Meade)
>   10. Re: NFAS t1 (Brian Meade)
>   11. Re: H245Interface & Related Processes (Daniel Pagan)
>   12. Re: Expressway - 3rd Party Border Recommendation (NateCCIE)
>   13. Re: UC integration with MS Office and Lync/MOC (Scott Voll)
>   14. Re: H245Interface & Related Processes (Daniel Pagan)
>   15. Re: Finisse -10.5 - cfadmin webpage doesnt open
>       (Jason Aarons (AM))
>   16. Re: Expressway - 3rd Party Border Recommendation (Josh Warcop)
>   17. Re: UC integration with MS Office and Lync/MOC (Josh Warcop)
>   18. Re: Expressway - 3rd Party Border Recommendation (Walenta, Philip)
>   19. Re: Jabber contact disappears consistently (Jason Aarons (AM))
>   20. CUSP High CPU utilization (Brian Palmer)
>   21. Re: H245Interface & Related Processes (Wes Sisk (wsisk))
>   22. Jabber 10.5.1 for Windows shows In a Meeting when he    isn't
>       (Jason Aarons (AM))
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Mon, 1 Dec 2014 18:23:59 +0000
> From: "Pawlowski, Adam" <[email protected]>
> To: "[email protected]" <[email protected]>
> Subject: [cisco-voip] Expressway - 3rd Party Border Recommendation
> Message-ID:
>       <aeb16c1bf994004e90a436c0cc62287f155fb...@mb-ls3.itorg.ad.buffalo.edu>
> Content-Type: text/plain; charset="us-ascii"
> 
> Afternoon all,
> 
>       Trying to get some opinion on how (if) you would put up a perimeter to 
> your UCM clusters to bring in 3rd party clients, softphones, etc, that are 
> SIP based and reside outside of your secured LAN? Most of our desktops are on 
> public addresses, not behind any particular hardware firewall, just software 
> on the host. I'm concerned that the host could be compromised, or as seen 
> with some soft clients, they just get harassed by driveby SIP/H.323 scans and 
> calls. 
> 
>       I haven't seen any great justification for trying to fence/proxy 
> connectivity to the UCM for Jabber, X-Lite, etc, to the cluster, but general 
> security practice is saying that if you can make it more secure, it is at 
> least worth looking into. 
> 
>       I've looked at trying to set the UBE up for proxy/passthrough 
> registrar, and this seems tedious because it doesn't proxy auth and requires 
> dial-peer configuration (making dual usage as a gateway cumbersome). I have 
> heard "use expressway" a few times but have no idea how that would work for 
> 3rd party SIP devices. Other than that, I spent a bit of time looking at 
> stuff from Edgewater, OpenSIPS, etc, but it is not clear to me if any of 
> these products are worth the trouble, and what the Cisco recommended way to 
> go about this is.
> 
>       Anyone have any experience or thought in this area? Is this a bad idea? 
> Anything to say about trying to secure potentially 'untrusted' connectivity 
> on a larger scale?
> 
> Regards,
> 
> Adam Pawlowski
> SUNYAB
> 
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Mon, 1 Dec 2014 13:51:18 -0500
> From: Brian Meade <[email protected]>
> To: "Pawlowski, Adam" <[email protected]>
> Cc: "[email protected]" <[email protected]>
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> Message-ID:
>       <CAGcuYh1dBMj6ij1hpo9j3NZ4U=5sde1-qcmk8wrtbzddfo4...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> I've done this before with a large Avaya setup.  We had all of the UC stuff
> in a separate VRF and all soft clients had to come through an SBC for
> registration.  We demoed Sipera and Acme.  Sipera got the job done cheaper,
> but Acme scaled much better for us.  I think CUCM supports Acme SBCs as
> well as an alternative to CUBE.
> 
> Brian
> 
> On Mon, Dec 1, 2014 at 1:23 PM, Pawlowski, Adam <[email protected]> wrote:
> 
> > Afternoon all,
> >
> >         Trying to get some opinion on how (if) you would put up a
> > perimeter to your UCM clusters to bring in 3rd party clients, softphones,
> > etc, that are SIP based and reside outside of your secured LAN? Most of our
> > desktops are on public addresses, not behind any particular hardware
> > firewall, just software on the host. I'm concerned that the host could be
> > compromised, or as seen with some soft clients, they just get harassed by
> > driveby SIP/H.323 scans and calls.
> >
> >         I haven't seen any great justification for trying to fence/proxy
> > connectivity to the UCM for Jabber, X-Lite, etc, to the cluster, but
> > general security practice is saying that if you can make it more secure, it
> > is at least worth looking into.
> >
> >         I've looked at trying to set the UBE up for proxy/passthrough
> > registrar, and this seems tedious because it doesn't proxy auth and
> > requires dial-peer configuration (making dual usage as a gateway
> > cumbersome). I have heard "use expressway" a few times but have no idea how
> > that would work for 3rd party SIP devices. Other than that, I spent a bit
> > of time looking at stuff from Edgewater, OpenSIPS, etc, but it is not clear
> > to me if any of these products are worth the trouble, and what the Cisco
> > recommended way to go about this is.
> >
> >         Anyone have any experience or thought in this area? Is this a bad
> > idea? Anything to say about trying to secure potentially 'untrusted'
> > connectivity on a larger scale?
> >
> > Regards,
> >
> > Adam Pawlowski
> > SUNYAB
> >
> > _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/c4b157f7/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 3
> Date: Mon, 1 Dec 2014 14:09:47 -0500
> From: Ryan Huff <[email protected]>
> To: "[email protected]" <[email protected]>
> Subject: [cisco-voip] Cisco Presence External Database Front End
>       Application
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> I have had a few requests from folks for help with a front-end GUI 
> application for the external PostgreSQL database that can be used with the 
> Cisco Presence and IM server.
> 
> Using something like PgAdmin is a great tool to use in an administrative 
> function but not very user friendly for the non tech. I have written a PHP 
> based application (developed on a LAMP stack). That is a great front-end GUI 
> for basic functionality. The application has 2 of the 4 features finished and 
> can be a great learning tool or a head start to writing/expanding your own 
> app.
> 
> Please download at: http://ryanthomashuff.com/downloads/
> 
> Let me know if you have any questions/need assistance getting it set up.
> 
> Thanks,
> 
> Ryan Huff
> ryanthomashuff.com
> CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist
>                                         
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/8507d271/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 4
> Date: Mon, 1 Dec 2014 20:53:52 +0000
> From: <[email protected]>
> To: <[email protected]>
> Subject: [cisco-voip] UC integration with MS Office and Lync/MOC
> Message-ID:
>       
> <255f57bb43f894468da6bd4f2f5dd1ceb77c2...@rw-s-exch03.net.its.l-3com.com>
>       
> Content-Type: text/plain; charset="us-ascii"
> 
> Hey Guys,
> 
>   We've been running Cisco UC integration version 8.X (CUCILYNC 8.6) for a 
> while now and it works great with MS Office Communicator R2 and Office 2010, 
> including the click to call add-in.  Our MS folks are looking to migrate 
> clients to MS Office 2013, but still use MOC R2.  I can't find any 
> product/solution that is compatible with MOC R2 as well as Office 2013 and 
> still provide click to call functionality.  Are there any options from Cisco 
> that we can use that will allow integration with MS Office Communicator R2 
> and Office 2013 that will provide call control and click to call too?  Also, 
> it seems the "new" CUCILYNC version is a totally separate window, instead of 
> an add-on section to the bottom, like it is with MOC R2.
> 
> Thanks,
> Bill
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/e6986472/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 5
> Date: Mon, 1 Dec 2014 22:12:46 +0000
> From: Daniel Pagan <[email protected]>
> To: "[email protected]" <[email protected]>
> Subject: [cisco-voip] H245Interface & Related Processes
> Message-ID:
>       <[email protected]>
> Content-Type: text/plain; charset="windows-1252"
> 
> Folks:
> 
> Hoping to get some insight on SDL process creation for H245...
> 
> Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
> Cluster-1 to Cluster-2... then Cluster-2 to Cluster-3. Cluster-3 sends the 
> H245 address & port info via H225 ALERTING to Cluster-2, which then sends its 
> own to Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
> Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
> processes are being created for the MSD/TCS exchange. According to SDL traces 
> on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
> message is "waitForTransportEstablishment". On Cluster-1, the H245Interface 
> process is never created according to SDL traces, so we never even reach the 
> opportunity for TCS media caps exchange. MXTimeout occurs shortly after.
> 
> Question is... For a node receiving an H245 address & port info via H225 (the 
> calling cluster...), is creation of the H245Interface and/or related H245 
> process dependent on CUCM *first* establishing the new, 2nd TCP socket with 
> the remote H.323 endpoint that advertised the H.245 port. In other words, at 
> an SDL level, is H245Interface created only after the 2nd TCP session is 
> successfully established at the transport level for H245 TCP communication? 
> Knowing this would help me assess the likelihood of the issue being related 
> to issues at the TCP level.
> 
> Thanks!
> 
> - Dan
> 
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/340f9a2f/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 6
> Date: Mon, 1 Dec 2014 22:32:32 +0000
> From: "Ryan Ratliff (rratliff)" <[email protected]>
> To: Daniel Pagan <[email protected]>
> Cc: cisco-voip voyp list <[email protected]>
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="windows-1252"
> 
> Short answer without confirming in the lab is yes, when I send you my H245 
> address I expect you to start a TCP connection to me on that port so we can 
> start H245.
> 
> 
> -Ryan
> 
> On Dec 1, 2014, at 5:12 PM, Daniel Pagan 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> Folks:
> 
> Hoping to get some insight on SDL process creation for H245?
> 
> Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
> Cluster-1 to Cluster-2? then Cluster-2 to Cluster-3. Cluster-3 sends the H245 
> address & port info via H225 ALERTING to Cluster-2, which then sends its own 
> to Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
> Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
> processes are being created for the MSD/TCS exchange. According to SDL traces 
> on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
> message is ?waitForTransportEstablishment?. On Cluster-1, the H245Interface 
> process is never created according to SDL traces, so we never even reach the 
> opportunity for TCS media caps exchange. MXTimeout occurs shortly after.
> 
> Question is? For a node receiving an H245 address & port info via H225 (the 
> calling cluster?), is creation of the H245Interface and/or related H245 
> process dependent on CUCM *first* establishing the new, 2nd TCP socket with 
> the remote H.323 endpoint that advertised the H.245 port. In other words, at 
> an SDL level, is H245Interface created only after the 2nd TCP session is 
> successfully established at the transport level for H245 TCP communication? 
> Knowing this would help me assess the likelihood of the issue being related 
> to issues at the TCP level.
> 
> Thanks!
> 
> - Dan
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/36a22c9a/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 7
> Date: Mon, 1 Dec 2014 15:37:49 -0700
> From: Charles Goldsmith <[email protected]>
> To: voip puck <[email protected]>
> Subject: [cisco-voip] NFAS t1
> Message-ID:
>       <cagm7t+budbxtzuftt6q75lzgq7_y8awex0mh-tj1izknw5s...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> We are seeing an odd error on a migration from a Nortel setup over to an
> h323 setup.
> 
> the error message after entering PRI 2 is
> 
> *crsvr1(config-controller)#pri-group timeslots 1-24 nfas_ backup nfas_int 1
> nfas_group 1*
> 
> *%The Primary-group is already defined*
> *%The first definition of the primary-group must be removed before the
> primary-group can be redefined*
> 
> Getting this after putting the timeslots command on the first pri, then
> seeing this error on the 2nd controller.
> 
> This is on 15.2 on a 3945.  I've done this before with no issues, but have
> never seen this error before.
> 
> Thanks
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/bf4b1b0c/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 8
> Date: Mon, 1 Dec 2014 15:47:53 -0700
> From: Charles Goldsmith <[email protected]>
> To: Brian Meade <[email protected]>
> Cc: voip puck <[email protected]>
> Subject: Re: [cisco-voip] NFAS t1
> Message-ID:
>       <cagm7t+b923dro3nmp-di5v8zgqdn0+fd2pwl5a3titxggvg...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> No, it's not, we cleared the pri-group's on both controllers by shutting
> down both voice-port and serial interfaces, removed the pri-group's, then
> added in correct config for NFAS.
> 
> Going to open a TAC case in a bit on it.
> 
> On Mon, Dec 1, 2014 at 3:46 PM, Brian Meade <[email protected]> wrote:
> 
> > Is the pri-group already configured on that controller?  If so, you'll
> > need to shut down the voice-port, the serial interface, and then do "no
> > pri-group" on the controller before re-defining the pri-group configuration.
> >
> > On Mon, Dec 1, 2014 at 5:37 PM, Charles Goldsmith <[email protected]>
> > wrote:
> >
> >> We are seeing an odd error on a migration from a Nortel setup over to an
> >> h323 setup.
> >>
> >> the error message after entering PRI 2 is
> >>
> >> *crsvr1(config-controller)#pri-group timeslots 1-24 nfas_ backup nfas_int
> >> 1 nfas_group 1*
> >>
> >> *%The Primary-group is already defined*
> >> *%The first definition of the primary-group must be removed before the
> >> primary-group can be redefined*
> >>
> >> Getting this after putting the timeslots command on the first pri, then
> >> seeing this error on the 2nd controller.
> >>
> >> This is on 15.2 on a 3945.  I've done this before with no issues, but
> >> have never seen this error before.
> >>
> >> Thanks
> >>
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> [email protected]
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/d2a4633f/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 9
> Date: Mon, 1 Dec 2014 17:50:51 -0500
> From: Brian Meade <[email protected]>
> To: "Ryan Ratliff (rratliff)" <[email protected]>
> Cc: cisco-voip voyp list <[email protected]>
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> Message-ID:
>       <CAGcuYh3mfSeJcYwd7SiX-ESAuOwBF0Cj5iUaAdn+PwDEo4=u...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> Also should be fairly easy to capture via a packet capture on Cluster 1 if
> this is easily reproducible for the call flow.
> 
> Brian
> 
> On Mon, Dec 1, 2014 at 5:32 PM, Ryan Ratliff (rratliff) <[email protected]>
> wrote:
> 
> >  Short answer without confirming in the lab is yes, when I send you my
> > H245 address I expect you to start a TCP connection to me on that port so
> > we can start H245.
> >
> >
> > -Ryan
> >
> >  On Dec 1, 2014, at 5:12 PM, Daniel Pagan <[email protected]> wrote:
> >
> >   Folks:
> >
> >
> >
> > Hoping to get some insight on SDL process creation for H245?
> >
> >
> >
> > Scenario is three CUCM clusters communicating over ICTs. Call is routed
> > from Cluster-1 to Cluster-2? then Cluster-2 to Cluster-3. Cluster-3 sends
> > the H245 address & port info via H225 ALERTING to Cluster-2, which then
> > sends its own to Cluster-1. Issue is Cluster-1 never establishes the H245
> > session with Cluster-2. The H245 address and port is received on Cluster-1
> > but no H245 processes are being created for the MSD/TCS exchange. According
> > to SDL traces on Cluster-2, the latest state of H245 on the node *
> > *sending** the ALERTING message is ?waitForTransportEstablishment?. On
> > Cluster-1, the H245Interface process is never created according to SDL
> > traces, so we never even reach the opportunity for TCS media caps exchange.
> > MXTimeout occurs shortly after.
> >
> >
> >
> > Question is? For a node receiving an H245 address & port info via H225
> > (the calling cluster?), is creation of the H245Interface and/or related
> > H245 process dependent on CUCM **first**** establishing the new, 2nd TCP
> > socket with the remote H.323 endpoint that advertised the H.245 port. In
> > other words, at an SDL level, is H245Interface created only after the 2nd
> > TCP session is successfully established at the transport level for H245 TCP
> > communication? Knowing this would help me assess the likelihood of the
> > issue being related to issues at the TCP level.
> >
> >
> >
> > Thanks!
> >
> >
> >
> > - Dan
> >  _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/b63e1625/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 10
> Date: Mon, 1 Dec 2014 17:46:15 -0500
> From: Brian Meade <[email protected]>
> To: Charles Goldsmith <[email protected]>
> Cc: voip puck <[email protected]>
> Subject: Re: [cisco-voip] NFAS t1
> Message-ID:
>       <CAGcuYh2ApRD5KZCbD5hTVN9mjVp6=njyr-1kkodj_5hxmv6...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> Is the pri-group already configured on that controller?  If so, you'll need
> to shut down the voice-port, the serial interface, and then do "no
> pri-group" on the controller before re-defining the pri-group configuration.
> 
> On Mon, Dec 1, 2014 at 5:37 PM, Charles Goldsmith <[email protected]>
> wrote:
> 
> > We are seeing an odd error on a migration from a Nortel setup over to an
> > h323 setup.
> >
> > the error message after entering PRI 2 is
> >
> > *crsvr1(config-controller)#pri-group timeslots 1-24 nfas_ backup nfas_int
> > 1 nfas_group 1*
> >
> > *%The Primary-group is already defined*
> > *%The first definition of the primary-group must be removed before the
> > primary-group can be redefined*
> >
> > Getting this after putting the timeslots command on the first pri, then
> > seeing this error on the 2nd controller.
> >
> > This is on 15.2 on a 3945.  I've done this before with no issues, but have
> > never seen this error before.
> >
> > Thanks
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/9a872a2c/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 11
> Date: Tue, 2 Dec 2014 00:51:38 +0000
> From: Daniel Pagan <[email protected]>
> To: "Ryan Ratliff (rratliff)" <[email protected]>
> Cc: cisco-voip voyp list <[email protected]>
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> Message-ID:
>       <[email protected]>
> Content-Type: text/plain; charset="windows-1252"
> 
> Thanks Ryan - that's what I was hoping to hear. I'll try to set this up in a 
> lab to confirm with some simple ACLs.
> 
> - Dan
> 
> From: Ryan Ratliff (rratliff) [mailto:[email protected]]
> Sent: Monday, December 01, 2014 5:33 PM
> To: Daniel Pagan
> Cc: cisco-voip voyp list
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> 
> Short answer without confirming in the lab is yes, when I send you my H245 
> address I expect you to start a TCP connection to me on that port so we can 
> start H245.
> 
> 
> -Ryan
> 
> On Dec 1, 2014, at 5:12 PM, Daniel Pagan 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> Folks:
> 
> Hoping to get some insight on SDL process creation for H245...
> 
> Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
> Cluster-1 to Cluster-2... then Cluster-2 to Cluster-3. Cluster-3 sends the 
> H245 address & port info via H225 ALERTING to Cluster-2, which then sends its 
> own to Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
> Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
> processes are being created for the MSD/TCS exchange. According to SDL traces 
> on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
> message is "waitForTransportEstablishment". On Cluster-1, the H245Interface 
> process is never created according to SDL traces, so we never even reach the 
> opportunity for TCS media caps exchange. MXTimeout occurs shortly after.
> 
> Question is... For a node receiving an H245 address & port info via H225 (the 
> calling cluster...), is creation of the H245Interface and/or related H245 
> process dependent on CUCM *first* establishing the new, 2nd TCP socket with 
> the remote H.323 endpoint that advertised the H.245 port. In other words, at 
> an SDL level, is H245Interface created only after the 2nd TCP session is 
> successfully established at the transport level for H245 TCP communication? 
> Knowing this would help me assess the likelihood of the issue being related 
> to issues at the TCP level.
> 
> Thanks!
> 
> - Dan
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/187fa055/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 12
> Date: Mon, 1 Dec 2014 12:18:27 -0700
> From: NateCCIE <[email protected]>
> To: "'Brian Meade'" <[email protected]>, "'Pawlowski, Adam'"
>       <[email protected]>
> Cc: [email protected]
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="utf-8"
> 
> Expressway is the first thought, then CUBE Lineside proxy would be where to 
> go for 3rd party.
> 
>  
> 
> https://ciscocollab.wordpress.com/2014/04/08/cube-sip-lineside-phone-vpn-configuration/
> 
>  
> 
>  
> 
> From: cisco-voip [mailto:[email protected]] On Behalf Of 
> Brian Meade
> Sent: Monday, December 1, 2014 11:51 AM
> To: Pawlowski, Adam
> Cc: [email protected]
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> 
>  
> 
> I've done this before with a large Avaya setup.  We had all of the UC stuff 
> in a separate VRF and all soft clients had to come through an SBC for 
> registration.  We demoed Sipera and Acme.  Sipera got the job done cheaper, 
> but Acme scaled much better for us.  I think CUCM supports Acme SBCs as well 
> as an alternative to CUBE.
> 
>  
> 
> Brian
> 
>  
> 
> On Mon, Dec 1, 2014 at 1:23 PM, Pawlowski, Adam <[email protected] 
> <mailto:[email protected]> > wrote:
> 
> Afternoon all,
> 
>         Trying to get some opinion on how (if) you would put up a perimeter 
> to your UCM clusters to bring in 3rd party clients, softphones, etc, that are 
> SIP based and reside outside of your secured LAN? Most of our desktops are on 
> public addresses, not behind any particular hardware firewall, just software 
> on the host. I'm concerned that the host could be compromised, or as seen 
> with some soft clients, they just get harassed by driveby SIP/H.323 scans and 
> calls.
> 
>         I haven't seen any great justification for trying to fence/proxy 
> connectivity to the UCM for Jabber, X-Lite, etc, to the cluster, but general 
> security practice is saying that if you can make it more secure, it is at 
> least worth looking into.
> 
>         I've looked at trying to set the UBE up for proxy/passthrough 
> registrar, and this seems tedious because it doesn't proxy auth and requires 
> dial-peer configuration (making dual usage as a gateway cumbersome). I have 
> heard "use expressway" a few times but have no idea how that would work for 
> 3rd party SIP devices. Other than that, I spent a bit of time looking at 
> stuff from Edgewater, OpenSIPS, etc, but it is not clear to me if any of 
> these products are worth the trouble, and what the Cisco recommended way to 
> go about this is.
> 
>         Anyone have any experience or thought in this area? Is this a bad 
> idea? Anything to say about trying to secure potentially 'untrusted' 
> connectivity on a larger scale?
> 
> Regards,
> 
> Adam Pawlowski
> SUNYAB
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected] <mailto:[email protected]> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
>  
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/c477a406/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 13
> Date: Mon, 1 Dec 2014 15:28:28 -0800
> From: Scott Voll <[email protected]>
> To: "[email protected]" <[email protected]>
> Cc: "[email protected]" <[email protected]>
> Subject: Re: [cisco-voip] UC integration with MS Office and Lync/MOC
> Message-ID:
>       <cahgd+39j08yh01dzfycrkdw3xyyfuparqcf5ntohyqcztvd...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> Correct.  The new cicilync is more like jabber. If you go to the o365 cloud
> it pretty much breaks everything. Good luck
> 
> Scott
> 
> On Monday, December 1, 2014, <[email protected]> wrote:
> 
> >  Hey Guys,
> >
> >
> >
> >   We?ve been running Cisco UC integration version 8.X (CUCILYNC 8.6) for a
> > while now and it works great with MS Office Communicator R2 and Office
> > 2010, including the click to call add-in.  Our MS folks are looking to
> > migrate clients to MS Office 2013, but still use MOC R2.  I can?t find any
> > product/solution that is compatible with MOC R2 as well as Office 2013 and
> > still provide click to call functionality.  Are there any options from
> > Cisco that we can use that will allow integration with MS Office
> > Communicator R2 and Office 2013 that will provide call control and click to
> > call too?  Also, it seems the ?new? CUCILYNC version is a totally separate
> > window, instead of an add-on section to the bottom, like it is with MOC R2.
> >
> >
> >
> > Thanks,
> >
> > Bill
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/cf1412ab/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 14
> Date: Tue, 2 Dec 2014 01:07:15 +0000
> From: Daniel Pagan <[email protected]>
> To: Brian Meade <[email protected]>, "Ryan Ratliff (rratliff)"
>       <[email protected]>
> Cc: cisco-voip voyp list <[email protected]>
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> Message-ID:
>       <[email protected]>
> Content-Type: text/plain; charset="utf-8"
> 
> Hey Brian ? hope you?re doing well. This is a difficult issue to reproduce so 
> a pcap would be tricky to obtain.
> 
> I?ll try and recreate the issue in a lab and see what results I get from an 
> SDL process creation standpoint.
> 
> Thanks!
> 
> - Dan
> 
> 
> From: [email protected] [mailto:[email protected]] On Behalf Of Brian Meade
> Sent: Monday, December 01, 2014 5:51 PM
> To: Ryan Ratliff (rratliff)
> Cc: Daniel Pagan; cisco-voip voyp list
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> 
> Also should be fairly easy to capture via a packet capture on Cluster 1 if 
> this is easily reproducible for the call flow.
> 
> Brian
> 
> On Mon, Dec 1, 2014 at 5:32 PM, Ryan Ratliff (rratliff) 
> <[email protected]<mailto:[email protected]>> wrote:
> Short answer without confirming in the lab is yes, when I send you my H245 
> address I expect you to start a TCP connection to me on that port so we can 
> start H245.
> 
> 
> -Ryan
> 
> On Dec 1, 2014, at 5:12 PM, Daniel Pagan 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> Folks:
> 
> Hoping to get some insight on SDL process creation for H245?
> 
> Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
> Cluster-1 to Cluster-2? then Cluster-2 to Cluster-3. Cluster-3 sends the H245 
> address & port info via H225 ALERTING to Cluster-2, which then sends its own 
> to Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
> Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
> processes are being created for the MSD/TCS exchange. According to SDL traces 
> on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
> message is ?waitForTransportEstablishment?. On Cluster-1, the H245Interface 
> process is never created according to SDL traces, so we never even reach the 
> opportunity for TCS media caps exchange. MXTimeout occurs shortly after.
> 
> Question is? For a node receiving an H245 address & port info via H225 (the 
> calling cluster?), is creation of the H245Interface and/or related H245 
> process dependent on CUCM *first* establishing the new, 2nd TCP socket with 
> the remote H.323 endpoint that advertised the H.245 port. In other words, at 
> an SDL level, is H245Interface created only after the 2nd TCP session is 
> successfully established at the transport level for H245 TCP communication? 
> Knowing this would help me assess the likelihood of the issue being related 
> to issues at the TCP level.
> 
> Thanks!
> 
> - Dan
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/2c663b1d/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 15
> Date: Tue, 2 Dec 2014 01:35:18 +0000
> From: "Jason Aarons (AM)" <[email protected]>
> To: 0703Manjunath <[email protected]>, "cisco-voip
>       ([email protected])" <[email protected]>
> Subject: Re: [cisco-voip] Finisse -10.5 - cfadmin webpage doesnt open
> Message-ID:
>       <2eb6888cfb98614ea7384beb9af8b38216c...@usispsvexdb03.na.didata.local>
> Content-Type: text/plain; charset="utf-8"
> 
> What OS/Browser versions?  If IE have you tried Compatibility mode?
> 
> http://docwiki.cisco.com/wiki/Unified_CCE_Software_Compatibility_Matrix_for_10.5%28x%29#Unified_CCE_Release_10.5..281.29_Supported_Browsers
> 
> 
> From: cisco-voip [mailto:[email protected]] On Behalf Of 
> 0703Manjunath
> Sent: Saturday, November 29, 2014 12:43 PM
> To: cisco-voip ([email protected])
> Subject: [cisco-voip] Finisse -10.5 - cfadmin webpage doesnt open
> 
> 
> hello,
> 
> 
> Iam stuck with  a issue on finesse  cfadmin  webpage doesnt open.
> 
> we are using UCCE 10.5 , agent webpage opens but admin page doesnt.
> 
> Does anyone have any clue about this issue????
> 
> --
> Cheers
> Manjunath
> 
> 
> itevomcid
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/1949f1be/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 16
> Date: Mon, 1 Dec 2014 21:51:38 -0500
> From: Josh Warcop <[email protected]>
> To: NateCCIE <[email protected]>, "'Brian Meade'" <[email protected]>,
>       "'Pawlowski, Adam'" <[email protected]>
> Cc: [email protected]
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="utf-8"
> 
> TAC opened 3 bugs on my behalf related to CUBE line-side SIP proxy. Not 
> including the documentation bugs that were opened.  CUBE in that fashion has 
> a few specific use cases and in my simple use case of replacing ASA 
> phone-proxy it didn't hold up. Expressway is your go to solution for Jabber 
> and TC endpoints and soon DX series.
> 
> Not saying CUBE proxy is terrible, but I would tread carefully down that path 
> and do plenty of testing.
> 
> Sent from my Windows Phone
> ________________________________
> From: NateCCIE<mailto:[email protected]>
> Sent: ?12/?1/?2014 7:58 PM
> To: 'Brian Meade'<mailto:[email protected]>; 'Pawlowski, 
> Adam'<mailto:[email protected]>
> Cc: [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> 
> Expressway is the first thought, then CUBE Lineside proxy would be where to 
> go for 3rd party.
> 
> 
> 
> https://ciscocollab.wordpress.com/2014/04/08/cube-sip-lineside-phone-vpn-configuration/
> 
> 
> 
> 
> 
> From: cisco-voip [mailto:[email protected]] On Behalf Of 
> Brian Meade
> Sent: Monday, December 1, 2014 11:51 AM
> To: Pawlowski, Adam
> Cc: [email protected]
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> 
> 
> 
> I've done this before with a large Avaya setup.  We had all of the UC stuff 
> in a separate VRF and all soft clients had to come through an SBC for 
> registration.  We demoed Sipera and Acme.  Sipera got the job done cheaper, 
> but Acme scaled much better for us.  I think CUCM supports Acme SBCs as well 
> as an alternative to CUBE.
> 
> 
> 
> Brian
> 
> 
> 
> On Mon, Dec 1, 2014 at 1:23 PM, Pawlowski, Adam <[email protected] 
> <mailto:[email protected]> > wrote:
> 
> Afternoon all,
> 
>         Trying to get some opinion on how (if) you would put up a perimeter 
> to your UCM clusters to bring in 3rd party clients, softphones, etc, that are 
> SIP based and reside outside of your secured LAN? Most of our desktops are on 
> public addresses, not behind any particular hardware firewall, just software 
> on the host. I'm concerned that the host could be compromised, or as seen 
> with some soft clients, they just get harassed by driveby SIP/H.323 scans and 
> calls.
> 
>         I haven't seen any great justification for trying to fence/proxy 
> connectivity to the UCM for Jabber, X-Lite, etc, to the cluster, but general 
> security practice is saying that if you can make it more secure, it is at 
> least worth looking into.
> 
>         I've looked at trying to set the UBE up for proxy/passthrough 
> registrar, and this seems tedious because it doesn't proxy auth and requires 
> dial-peer configuration (making dual usage as a gateway cumbersome). I have 
> heard "use expressway" a few times but have no idea how that would work for 
> 3rd party SIP devices. Other than that, I spent a bit of time looking at 
> stuff from Edgewater, OpenSIPS, etc, but it is not clear to me if any of 
> these products are worth the trouble, and what the Cisco recommended way to 
> go about this is.
> 
>         Anyone have any experience or thought in this area? Is this a bad 
> idea? Anything to say about trying to secure potentially 'untrusted' 
> connectivity on a larger scale?
> 
> Regards,
> 
> Adam Pawlowski
> SUNYAB
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected] <mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/f960fe3e/attachment-0001.html>
> -------------- next part --------------
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> ------------------------------
> 
> Message: 17
> Date: Mon, 1 Dec 2014 22:01:28 -0500
> From: Josh Warcop <[email protected]>
> To: "[email protected]" <[email protected]>,
>       <[email protected]>
> Subject: Re: [cisco-voip] UC integration with MS Office and Lync/MOC
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="utf-8"
> 
> I'm first coming at this from a Microsoft perspective. Have you moved to Lync 
> Server 2013 and running OCS for CUCILYNC backward compatibility?
> 
> MS Office 2013 includes the Lync 2013 client so you would have to suppress 
> that part of the upgrade.
> 
> You're not finding products compatible with both because Microsoft forked 
> that between OCS and Lync with the Office 2007 to Office 2013 transition.
> 
> My recommendation is to drop CUCILYNC.
> 
> Sent from my Windows Phone
> ________________________________
> From: [email protected]<mailto:[email protected]>
> Sent: ?12/?1/?2014 3:57 PM
> To: [email protected]<mailto:[email protected]>
> Subject: [cisco-voip] UC integration with MS Office and Lync/MOC
> 
> Hey Guys,
> 
>   We've been running Cisco UC integration version 8.X (CUCILYNC 8.6) for a 
> while now and it works great with MS Office Communicator R2 and Office 2010, 
> including the click to call add-in.  Our MS folks are looking to migrate 
> clients to MS Office 2013, but still use MOC R2.  I can't find any 
> product/solution that is compatible with MOC R2 as well as Office 2013 and 
> still provide click to call functionality.  Are there any options from Cisco 
> that we can use that will allow integration with MS Office Communicator R2 
> and Office 2013 that will provide call control and click to call too?  Also, 
> it seems the "new" CUCILYNC version is a totally separate window, instead of 
> an add-on section to the bottom, like it is with MOC R2.
> 
> Thanks,
> Bill
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/3659c5f7/attachment-0001.html>
> -------------- next part --------------
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> ------------------------------
> 
> Message: 18
> Date: Mon, 1 Dec 2014 19:29:44 -0800
> From: "Walenta, Philip" <[email protected]>
> To: Josh Warcop <[email protected]>
> Cc: "[email protected]" <[email protected]>,
>       "Pawlowski, Adam" <[email protected]>
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="utf-8"
> 
> I would second this opinion.  I've done a lot of work with CUBE and it does 
> not handle phone proxy well at all (or much when it comes to endpoint auth 
> situations). VCS is your best best.
> 
> Sent from my iPhone
> 
> On Dec 1, 2014, at 8:55 PM, Josh Warcop 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> TAC opened 3 bugs on my behalf related to CUBE line-side SIP proxy. Not 
> including the documentation bugs that were opened.  CUBE in that fashion has 
> a few specific use cases and in my simple use case of replacing ASA 
> phone-proxy it didn't hold up. Expressway is your go to solution for Jabber 
> and TC endpoints and soon DX series.
> 
> Not saying CUBE proxy is terrible, but I would tread carefully down that path 
> and do plenty of testing.
> 
> Sent from my Windows Phone
> ________________________________
> From: NateCCIE<mailto:[email protected]>
> Sent: ?12/?1/?2014 7:58 PM
> To: 'Brian Meade'<mailto:[email protected]>; 'Pawlowski, 
> Adam'<mailto:[email protected]>
> Cc: [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> 
> 
> Expressway is the first thought, then CUBE Lineside proxy would be where to 
> go for 3rd party.
> 
> 
> 
> https://ciscocollab.wordpress.com/2014/04/08/cube-sip-lineside-phone-vpn-configuration/
> 
> 
> 
> 
> 
> From: cisco-voip [mailto:[email protected]] On Behalf Of 
> Brian Meade
> Sent: Monday, December 1, 2014 11:51 AM
> To: Pawlowski, Adam
> Cc: [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Expressway - 3rd Party Border Recommendation
> 
> 
> 
> I've done this before with a large Avaya setup.  We had all of the UC stuff 
> in a separate VRF and all soft clients had to come through an SBC for 
> registration.  We demoed Sipera and Acme.  Sipera got the job done cheaper, 
> but Acme scaled much better for us.  I think CUCM supports Acme SBCs as well 
> as an alternative to CUBE.
> 
> 
> 
> Brian
> 
> 
> 
> On Mon, Dec 1, 2014 at 1:23 PM, Pawlowski, Adam 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> Afternoon all,
> 
>         Trying to get some opinion on how (if) you would put up a perimeter 
> to your UCM clusters to bring in 3rd party clients, softphones, etc, that are 
> SIP based and reside outside of your secured LAN? Most of our desktops are on 
> public addresses, not behind any particular hardware firewall, just software 
> on the host. I'm concerned that the host could be compromised, or as seen 
> with some soft clients, they just get harassed by driveby SIP/H.323 scans and 
> calls.
> 
>         I haven't seen any great justification for trying to fence/proxy 
> connectivity to the UCM for Jabber, X-Lite, etc, to the cluster, but general 
> security practice is saying that if you can make it more secure, it is at 
> least worth looking into.
> 
>         I've looked at trying to set the UBE up for proxy/passthrough 
> registrar, and this seems tedious because it doesn't proxy auth and requires 
> dial-peer configuration (making dual usage as a gateway cumbersome). I have 
> heard "use expressway" a few times but have no idea how that would work for 
> 3rd party SIP devices. Other than that, I spent a bit of time looking at 
> stuff from Edgewater, OpenSIPS, etc, but it is not clear to me if any of 
> these products are worth the trouble, and what the Cisco recommended way to 
> go about this is.
> 
>         Anyone have any experience or thought in this area? Is this a bad 
> idea? Anything to say about trying to secure potentially 'untrusted' 
> connectivity on a larger scale?
> 
> Regards,
> 
> Adam Pawlowski
> SUNYAB
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141201/c39f163c/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 19
> Date: Tue, 2 Dec 2014 13:21:02 +0000
> From: "Jason Aarons (AM)" <[email protected]>
> To: Josh Warcop <[email protected]>, Ryan Huff <[email protected]>,
>       Pavan K <[email protected]>, Leslie Meade <[email protected]>
> Cc: "cisco-voip \([email protected]\)"
>       <[email protected]>
> Subject: Re: [cisco-voip] Jabber contact disappears consistently
> Message-ID:
>       <2eb6888cfb98614ea7384beb9af8b38216c...@usispsvexdb03.na.didata.local>
> Content-Type: text/plain; charset="utf-8"
> 
> Yes URI Dialing works great and is kind of neat.  What we are discussing is a 
> setting in IMP to select DirectoryURI as the Jabber ID (JID) address scheme 
> for Presence/Chat.  Has nothing to do with URI Dialing.
> 
> CSCuo95266 - Jabber Windows and directory URI address scheme in IMP 10.x -doc 
> defect
> Symptom:
> Documentation defect for Jabber for Windows support with  directory URI 
> address scheme which was introduced in  IMP 10.x
> 
> Cisco Jabber for Windows - 9.7.1
> IM and Presence - 10.x
> 
> Conditions:
> Jabber for Windows 9.7 does not support directory URI address scheme which 
> was introduced in IM and Presence 10.x. Currently Jabber for windows 10.5 
> does not support enhanced directory URI,
>  10.6 will support use of a single domain with directoryURI. <<<<THIS HAS NOT 
> BEEN COMMITTED PER BU
> 
> But Cisco J4W 9.7 installation guide incorrectly states that the feature is 
> supported.
> 
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/Windows/9_7/JABW_BK_C4C679C9_00_cisco-jabber-for-windows-97/JABW_BK_C4C679C9_00_cisco-jabber-for-windows-97_chapter_0111.html#CJAB_TK_P8652662_00
> 
> 
> 
> 
> From: Ryan Huff<mailto:[email protected]>
> Sent: ?Wednesday?, ?November? ?26?, ?2014 ?11?:?59? ?AM
> To: Pavan K<mailto:[email protected]>, Leslie 
> Meade<mailto:[email protected]>
> Cc: cisco-voip ([email protected])<mailto:[email protected]>
> 
> I call BS!
> 
> I am running a fully AD integrated 9.1.2 CUCM with IM&Presence 9.1.2 and I 
> have the Cisco Jabber for Windows 10.5 client deployed and I have URI Dialing 
> running in the Jabber client, ALL. DAY. LONG and it works flawlessly.
> 
> In the deployment docs, it says to use EnableSIPUriDialing in the 
> jabber-config.xml HOWEVER; if you look at the PRT logs generated from one of 
> the 10.5 Jabber clients I bet you it is looking for the EnableSIPURIDialing 
> node and not the EnableSIPUriDialing node.
> 
> Notice the casing difference in URI Vs. Uri ..... I couldn't get URI in the 
> client for weeks, got the same junk answer from TAC as well .... then I 
> troll'ed the PRT logs and found that to be my issue, a flippin' casing issue.
> 
> 
> ________________________________
> Date: Wed, 26 Nov 2014 09:09:50 -0600
> From: [email protected]<mailto:[email protected]>
> To: [email protected]<mailto:[email protected]>
> CC: [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Jabber contact disappears consistently
> Just to close this out.
> 
> Official word from Cisco even though every piece of Jabber doc says its 
> supported in J4W 10.5
> ==
> 
> Currently Directory URI is only supported on the server.  The client has not 
> yet been updated to fully support this.  So no, Jabber 10.5 does NOT support 
> directory URI
> 
> ==
> 
> 
> On Sat, Nov 22, 2014 at 2:15 PM, Leslie Meade 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> I also have seen the same issue, and put everyone onto the same server
> 
> 
> 
> Leslie Meade
> 
> 
> 
> ..................................................................
> Mobile:778.228.4339 | Main: 604.676.5239
> Email: [email protected]<mailto:[email protected]>
> 
> 
> 
> From: cisco-voip 
> [mailto:[email protected]<mailto:[email protected]>]
>  On Behalf Of Jason Aarons (AM)
> Sent: Saturday, November 22, 2014 11:33 AM
> To: Josh Warcop; Pavan K; 
> [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Jabber contact disappears consistently
> 
> 
> 
> I complete agree, put all Jabberusers on single server.  Saw the same bugs in 
> 10x.
> 
> 
> 
> From: cisco-voip [mailto:[email protected]] On Behalf Of 
> Josh Warcop
> Sent: Friday, November 21, 2014 3:48 PM
> To: Pavan K; [email protected]<mailto:[email protected]>
> Subject: Re: [cisco-voip] Jabber contact disappears consistently
> 
> 
> 
> 
> 
> Do to the bugs I've seen I have been pinning all users to a single node in 
> the cluster and just having HA failover.
> 
> 
> ________________________________
> 
> Date: Fri, 21 Nov 2014 11:31:39 -0600
> From: [email protected]<mailto:[email protected]>
> To: [email protected]<mailto:[email protected]>
> Subject: [cisco-voip] Jabber contact disappears consistently
> We have an interesting problem on a new jabber deployment that has us 
> stumped. Wonder if anybody else saw this.
> Two jabber servers with ha enabled and balanced users. Using 10.5su1 for 
> jabber windows and ucm/imp.
> Leveraging sip directory uri as the IM scheme due to a multi forest 
> environment with duplicate Samaccountnames across domains.
> UserA contact list has userB on it. Folks can im each other without any 
> problem. If we move userB from his imp server to another server in the same 
> subcluster, userB disappears from userA's contact list.
> Repeatable across multiple users with different userA and userB and happens 
> every time regardless of moving them from server1 to server2 or vice versa.
> Using router to router communication between nodes and default jabber-config.
> 
> Any ideas ?
> Have a TAC case open but its going nowhere.
> 
> _______________________________________________ cisco-voip mailing list 
> [email protected]<mailto:[email protected]> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> itevomcid
> 
> 
> 
> --
> - Pavan
> 
> _______________________________________________ cisco-voip mailing list 
> [email protected]<mailto:[email protected]> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/b2fc815c/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 20
> Date: Tue, 02 Dec 2014 15:01:06 +0000
> From: "Brian Palmer" <[email protected]>
> To: [email protected]
> Subject: [cisco-voip] CUSP High CPU utilization
> Message-ID: <em8c62e2a7-813a-46ab-b2cc-e665336c1c68@praetorian>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
> 
> Got a customer with a CUSP and both sides seem to be pegging very very 
> high on the cpu usage.  So high I believe it is interfering with call 
> processing.  The vendor that put this new cluster did not follow best 
> practices in general so there are all kinds of things going on like 
> criss-crossing of media between the two geographically separated sides.  
> My real problem is that the CUSP module is much harder to diagnose than 
> say a regular voice gateway due to a lack of commands to figure out what 
> is pegging the cpu so bad.  Anybody have any good ideas?  I can say the 
> licensing is woefully inadequate and looks to be based on average CPS 
> and not peak though I don't think that would have an impact on cpu 
> usage.
> 
> 
> thanks
> 
> Brian
> 
> -- 
> *The information contained in this electronic transmission and any 
> attachments hereto may be considered proprietary and confidential. 
> Unauthorized distribution of this material to anyone other than the 
> addressed is prohibited. Any disclosure, distribution or use of this 
> transmission for any reason other than their intended purpose is 
> prohibited.*
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/a975fd1a/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 21
> Date: Tue, 2 Dec 2014 16:29:04 +0000
> From: "Wes Sisk (wsisk)" <[email protected]>
> To: Daniel Pagan <[email protected]>
> Cc: "[email protected]" <[email protected]>
> Subject: Re: [cisco-voip] H245Interface & Related Processes
> Message-ID: <[email protected]>
> Content-Type: text/plain; charset="windows-1252"
> 
> Hmm,
> 
> These should help:
> CSCsm26337    need clear messages for h225 tcp session failure
> 
> In versions with fix h225 session setup is clearly indicated in cm sdi traces
> with one of these messages:
> H225Cdpc(%07d)::requestConnect_TcpStartSessionErr: H225 Tcp Start Session
> request failed
> 
> H225 Call is aborted as no response received for H225 Tcp Start Session 
> Request
> within configured H225TcpTimer value
> 
> h225 session abort due to RST or FIN is clearly indicated in cm sdi traces 
> with:
> H225 Tcp session terminated abnormally
> 
> 
> With fix for this defect TCP session aborts will be reported by one of the 
> following statements in CallManager SDI trace:
> requestConnect_TcpStartSessionErr: H225 Tcp Start Session request failed
> requestConnect_H225TcpTimer: H225 Call is aborted as no response received for 
> H225 Tcp Start Session Request within configured H225TcpTimer value
> call_initiated1_TcpStopSessionInd: H225 Tcp session terminated abnormally
> overlap_sending2_TcpStopSessionInd: H225 Tcp session terminated abnormally
> outgoing_call_proceeding3_TcpStopSessionInd: H225 Tcp session terminated 
> abnormally
> call_delivered4_TcpStopSessionInd: H225 Tcp session terminated abnormally
> call_present6_TcpStopSessionInd: H225 Tcp session terminated abnormally
> incoming_call_proceeding9_TcpStopSessionInd: H225 Tcp session terminated 
> abnormally
> active10_TcpStopSessionInd: H225 Tcp session terminated abnormally
> await_ann_complete_TcpStopSessionInd: H225 Tcp session terminated abnormally
> active10a_TcpStopSessionInd: H225 Tcp session terminated abnormally
> active10b_TcpStopSessionInd: H225 Tcp session terminated abnormally
> GKRasARQH225Setup_TcpStopSessionInd: H225 Tcp session terminated abnormally
> GKRasCcSetupRequestConnect_TcpStartSessionErr(%d, %d): TcpStartSessionErr 
> from IP=%s
> GKRasCcSetupRequestConnect_TcpStartSessionErr: H225 Tcp Start Session request 
> failed
> GKRasCcSetupRequestConnect_H225TcpTimer: H225 Call is aborted as no response 
> received for H225 Tcp Start Session Request within configured H225TcpTimer 
> value
> overlap_receiving25_TcpStopSessionInd: H225 Tcp session terminated abnormally
> wait_for_disconn_kluge_TcpStopSessionInd: H225 Tcp session terminated 
> abnormally
> paused_TcpStopSessionInd: H225 Tcp session terminated abnormally
> 
> 
> 
> CSCsm26355    need clear messages for h245 tcp session failure
> h.245 tcp session setup failure:
> TCP ERROR: H245ListenReq or H245ConnectReq failure, or received SdlCloseInd
> from H245Handler, Perform cleanup of H245 Session
> 
> established h.245 session experiences TCP keepalive timeout:
> TranslateAndTransport(%d)::wait_SdlCloseInd - ERROR: H245 signaling connection
> aborted
> 
> established h.245 session receives unexpected TCP FIN or RST:
> TranslateAndTransport(%d)::wait_SdlCloseInd - ERROR: H245 signaling connection
> aborted
> 
> 
> 
> 
> Otherwise, consider enabling additional SDL trace flags like ?enable network 
> *? and ?enable SDL TCP event trace?.  However, these will cause significant 
> additional trace lines (one or more SDL signal per TCP segment received).
> 
> 
> I believe one of the additional traces above will get you insight to socket 
> requests down to the network layer in UCM.
> 
> -Wes
> 
> 
> 
> On Dec 1, 2014, at 5:12 PM, Daniel Pagan 
> <[email protected]<mailto:[email protected]>> wrote:
> 
> Folks:
> 
> Hoping to get some insight on SDL process creation for H245?
> 
> Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
> Cluster-1 to Cluster-2? then Cluster-2 to Cluster-3. Cluster-3 sends the H245 
> address & port info via H225 ALERTING to Cluster-2, which then sends its own 
> to Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
> Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
> processes are being created for the MSD/TCS exchange. According to SDL traces 
> on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
> message is ?waitForTransportEstablishment?. On Cluster-1, the H245Interface 
> process is never created according to SDL traces, so we never even reach the 
> opportunity for TCS media caps exchange. MXTimeout occurs shortly after.
> 
> Question is? For a node receiving an H245 address & port info via H225 (the 
> calling cluster?), is creation of the H245Interface and/or related H245 
> process dependent on CUCM *first* establishing the new, 2nd TCP socket with 
> the remote H.323 endpoint that advertised the H.245 port. In other words, at 
> an SDL level, is H245Interface created only after the 2nd TCP session is 
> successfully established at the transport level for H245 TCP communication? 
> Knowing this would help me assess the likelihood of the issue being related 
> to issues at the TCP level.
> 
> Thanks!
> 
> - Dan
> _______________________________________________
> cisco-voip mailing list
> [email protected]<mailto:[email protected]>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/d3dc1152/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 22
> Date: Tue, 2 Dec 2014 16:31:49 +0000
> From: "Jason Aarons (AM)" <[email protected]>
> To: "cisco-voip ([email protected])"
>       <[email protected]>
> Subject: [cisco-voip] Jabber 10.5.1 for Windows shows In a Meeting
>       when he isn't
> Message-ID:
>       <2eb6888cfb98614ea7384beb9af8b382176...@usispsvexdb03.na.didata.local>
> Content-Type: text/plain; charset="windows-1252"
> 
> We have a user who's Jabber for Windows 10.5.1 client shows "In a Meeting" 
> when his calendar is clear. I looked in outlook if he has multiple calendars 
> which he did not, when he logs into Jabber his status shows as "available" 
> for around 15 seconds before changing to "In a meeting". If I clear the local 
> Jabber CSF folders it still displays the same behavior.
> 
> IMP 10.5.1
> 
> 
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> <https://puck.nether.net/pipermail/cisco-voip/attachments/20141202/4357a4b1/attachment-0001.html>
> 
> ------------------------------
> 
> Subject: Digest Footer
> 
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> ------------------------------
> 
> End of cisco-voip Digest, Vol 134, Issue 2
> ******************************************
                                          
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip

Reply via email to