Wes is right, on Version 6.X DNS and NTP are very sensitive, during an upgrade/install you want to have those configured.<br>I've had the error before and opened a TAC case. I had to rebuild my server from scratch and i had to configure DNS and NTP for the install to complete successfully<br>
<br><div class="gmail_quote">2009/6/25 Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
In that case, yep, fully agree. DNS and NTP are quick ways to torpedo
an install.
/Wes
On Thursday, June 25, 2009 2:04:08 PM, Robert Knapp
wrote:
> At this point I'm reimaging the pub. The pub install had issues like access to DNS and NTP. Figured it would better to have clean install of pub.
>
> Robert Knapp
>
> Sent from a shiny magical device with predictive typing.
>
> On Jun 25, 2009, at 2:00 PM, "Wes Sisk" > wrote:
>
> Hmm, A few hits on this error. A few things to check:
>
> A. what is complete sequence of upgrade? Others with this error had success with this order of events:
> 1. upgrade pub, do not reboot
> 2. upgrade sub, do not reboot
> 3. reboot pub
> 4. reboot sub
>
> B. CSCsm21623 CUCM upgrade from 5.1 to 6.1 on Sub fails when hostname case mismatch
> Appears triggered when hostname case does not match case in process node table.
>
> to get name from database use admin cli:
> run sql select name from processnode
>
> to get platform name use admin cli:
> show status
>
>
> C. Subscriber install fails attempting to restore ontape backup data copied down from publisher.
> Either backup file is corrupt on publisher, sftp transfer over network fails, or file is corrupted on disk on subscriber. Initiating a new backup on the publisher may generate a new ontape file to be used for upgrade.
>
> D. Servers were originally installed and locales added. Failure occurred and servers had to be reinstalled and restored from backup/DRS/DRF. After reinstall Locales are not re-installed. Next upgrade fails due to files referenced in the database(from backup) missing from filesystem(from install).
>
>
> E. Subscriber runs out of disk space. Check free space in the active, inactive, and common partitions. If a large number of locales are installed look out for CSCsz58138.
>
>
> The install.log file from upgrade should display more information about the nature of failure if you have the fix for
> CSCso46012 Better error reporting on ontape backup and ontape restore failures
>
> Regards,
> Wes
>
>
> On Thursday, June 25, 2009 12:22:39 PM, Robert Knapp <<a href="mailto:robert.knapp@spanlink.com">robert.knapp@spanlink.com</a>> wrote:
>
> Having "issues" with upgrade.
> the CLI displays error: syslogd: /var/log/active/platform/log/authneticatefile.log : No such file or directory
>
> I am upgrading via the a browser:
> the log says such items as
> exception ontaperestore
> and
> raise Exception, ("exception caught during ontape restore [%s]" % msg)|
> 06/25/2009 12:14:38 CCMInstall|Internal Error, File:instMain.c:1403, Function: handlePhase(), Failed to exec
>
> I have reboot the pub and sub, any suggestions?
>
> Thanks,
>
> Robert Knapp
>
> ________________________________________
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of <a href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a> [<a href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a>]
> Sent: Thursday, June 25, 2009 12:00 PM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: cisco-voip Digest, Vol 68, Issue 23
>
> Send cisco-voip mailing list submissions to
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
>
> To subscribe or unsubscribe via the World Wide Web, visit
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> or, via email, send a message with subject or body 'help' to
> <a href="mailto:cisco-voip-request@puck.nether.net">cisco-voip-request@puck.nether.net</a>
>
> You can reach the person managing the list at
> <a href="mailto:cisco-voip-owner@puck.nether.net">cisco-voip-owner@puck.nether.net</a>
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of cisco-voip digest..."
>
>
> Today's Topics:
>
> 1. Has Anyone Run A Primary Unity Server on a Physical Server
> and the Failover on a Virtual Server? (Miller, Steve)
> 2. create a software CFB that got deleted (Tim Frazee)
> 3. Call Manager 5.1.1a install file? (Erick Bergquist)
> 4. Re: Has Anyone Run A Primary Unity Server on a Physical
> Server and the Failover on a Virtual Server? (Paul)
> 5. Re: Call Manager 5.1.1a install file? (Jason Burns)
> 6. Re: Has Anyone Run A Primary Unity Server on aPhysical Server
> and the Failover on a Virtual Server? (Jason Aarons (US))
> 7. Re: FW: CME web access disable (Nick Matthews)
> 8. Re: FW: CME web access disable (Ahmed Elnagar)
> 9. PRI Protocol NAT1, NAT2, custom? (Jeff Ruttman)
> 10. Re: PRI Protocol NAT1, NAT2, custom? (Matt Slaga (US))
> 11. Re: PRI Protocol NAT1, NAT2, custom? (Jeff Ruttman)
> 12. Re: PRI Protocol NAT1, NAT2, custom? (Matt Slaga (US))
> 13. Re: create a software CFB that got deleted (Peter Slow)
> 14. Re: destination-pattern "T" question (Dew Swen)
> 15. Does Unity Connection 7.1.2a support SIP RFC 2833 for DTMF
> (Jason Aarons (US))
> 16. Re: Multicast MoH Delay (Tony Underwood)
> 17. Re: Multicast MoH Delay (Daniel)
> 18. SIP Route Pattern (Jake Doe)
> 19. TAPS and UCCX 5.0.2 ? (Jason Aarons (US))
> 20. Re: Does Unity Connection 7.1.2a support SIP RFC 2833 for
> DTMF (Adam Frankel)
> 21. Re: Call Manager 5.1.1a install file? (Erick Bergquist)
> 22. T.37 Fax Redundancy (ciscozest)
> 23. Re: Multicast MoH Delay (Tony Underwood)
> 24. One User Cannot Be Dialed By Name (Miller, Steve)
> 25. Re: One User Cannot Be Dialed By Name (Cristobal Priego)
> 26. Re: TAPS and UCCX 5.0.2 ? (Dustin S Fowler)
> 27. Re: Does Unity Connection 7.1.2a support SIP RFC 2833 for
> DTMF (Mark Holloway)
> 28. Show Saved Enterprise Data in CAD (ROJAS, Mario)
> 29. Re: destination-pattern "T" question (Mehmet Turunc)
> 30. Re: Show Saved Enterprise Data in CAD (Beck, Christopher)
> 31. Slow to connect calls (Jeff Ruttman)
> 32. VoicemailQueuing (Voice Noob)
> 33. Re: Slow to connect calls (Ian MacKinnon)
> 34. TAC confirms incorrect filename on CCO (<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 24 Jun 2009 13:20:31 -0400
> From: "Miller, Steve" <MillerS@DicksteinShapiro.COM>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a
> Physical Server and the Failover on a Virtual Server?
> Message-ID: <<a href="mailto:418329B7ED67E64BBD2BAA97078D70D001C62105@DCEX2.DSMO.COM">418329B7ED67E64BBD2BAA97078D70D001C62105@DCEX2.DSMO.COM</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> We are looking to run Unity 7.0.2 on Server 2003. I am looking at a
> number of different scenarios, but I wanted to get a feel for whether
> this idea was totally crazy or just a little crazy.
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> <a href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</a>
>
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
> named above. This communication may contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for delivering this confidential
> communication to the intended recipient, you have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
> is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
> received, or sent on its network. If you have received this confidential communication in error, please notify the
> sender immediately by reply e-mail message and permanently delete the original message.
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> To reply to our email administrator directly, send an email to <a href="mailto:postmaster@dicksteinshapiro.com">postmaster@dicksteinshapiro.com</a>
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> Dickstein Shapiro LLP
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>
> ==============================================================================
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>
> ------------------------------
>
> Message: 2
> Date: Wed, 24 Jun 2009 12:43:35 -0500
> From: Tim Frazee <<a href="mailto:tfrazee@gmail.com">tfrazee@gmail.com</a>>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] create a software CFB that got deleted
> Message-ID:
> <<a href="mailto:30ce418a0906241043w2acc572fi4c612471267405b7@mail.gmail.com">30ce418a0906241043w2acc572fi4c612471267405b7@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> hey all,
>
> Running CUCM 7.0(2a) and somehow my software CFB (the one that runs on the
> server, from the IPVM service) got deleted. How do I recreate the CFB on the
> server?
>
> Adding a software CFB is not a choice in the drop down menu when I try to
> just rebuild it.
>
> Any ideas?
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>
> ------------------------------
>
> Message: 3
> Date: Wed, 24 Jun 2009 13:06:07 -0500
> From: Erick Bergquist <<a href="mailto:erickbee@gmail.com">erickbee@gmail.com</a>>
> To: cisco-voip mailinglist <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] Call Manager 5.1.1a install file?
> Message-ID:
> <<a href="mailto:f4445faf0906241106q6f3c2ddbjb7f4f78be34357ce@mail.gmail.com">f4445faf0906241106q6f3c2ddbjb7f4f78be34357ce@mail.gmail.com</a>>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Does anyone recall what the install file / version is for CUCM 5.1.1a?
> is it 3000-4?
>
> Thanks, Erick
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 24 Jun 2009 10:24:20 -0700 (PDT)
> From: Paul <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>>
> To: "Miller, Steve" <MillerS@DicksteinShapiro.COM>,
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on a
> Physical Server and the Failover on a Virtual Server?
> Message-ID: <<a href="mailto:851685.56437.qm@web111314.mail.gq1.yahoo.com">851685.56437.qm@web111314.mail.gq1.yahoo.com</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> First of all, is it supported?
>
> If it's not supported and you're planning on using it in a production environment, then yes you are crazy.
>
>
>
>
> ________________________________
> From: "Miller, Steve" <MillerS@DicksteinShapiro.COM>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Sent: Wednesday, June 24, 2009 1:20:31 PM
> Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a Physical Server and the Failover on a Virtual Server?
>
>
> We are looking to
> run Unity 7.0.2 on Server 2003. I am looking at a number of different
> scenarios, but I wanted to get a feel for whether this idea was totally crazy or
> just a little crazy.
>
> Steve Miller
> Telecom Engineer
> Dickstein
> Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370|
> Fax (202) 330-5607
> <a href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</a>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
> named above. This communication may contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for delivering this confidential
> communication to the intended recipient, you have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
> is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
> received, or sent on its network. If you have received this confidential communication in error, please notify the
> sender immediately by reply e-mail message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to <a href="mailto:postmaster@dicksteinshapiro.com">postmaster@dicksteinshapiro.com</a>
>
> Dickstein Shapiro LLP
> <a href="http://www.DicksteinShapiro.com">http://www.DicksteinShapiro.com</a>
>
> ==============================================================================
>
>
> __________________________________________________
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>
> ------------------------------
>
> Message: 5
> Date: Wed, 24 Jun 2009 14:28:32 -0400
> From: Jason Burns <<a href="mailto:burns.jason@gmail.com">burns.jason@gmail.com</a>>
> To: Erick Bergquist <<a href="mailto:erickbee@gmail.com">erickbee@gmail.com</a>>
> Cc: cisco-voip mailinglist <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Call Manager 5.1.1a install file?
> Message-ID:
> <<a href="mailto:78d9bfc20906241128u76aa5207pc5bec4e836ce6b0e@mail.gmail.com">78d9bfc20906241128u76aa5207pc5bec4e836ce6b0e@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> That would probably be 5.1.1.2000-1 or 2.
>
> On Wed, Jun 24, 2009 at 2:06 PM, Erick Bergquist <<a href="mailto:erickbee@gmail.com">erickbee@gmail.com</a>> wrote:
>
>
>
> Does anyone recall what the install file / version is for CUCM 5.1.1a?
> is it 3000-4?
>
> Thanks, Erick
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
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>
> ------------------------------
>
> Message: 6
> Date: Wed, 24 Jun 2009 14:39:05 -0400
> From: "Jason Aarons (US)" <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>
> To: "Paul" <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>>, "Miller, Steve"
> <MillerS@DicksteinShapiro.COM>, <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on
> aPhysical Server and the Failover on a Virtual Server?
> Message-ID:
> <C1FE15183DA37645BC0633BC604E44F00F48FA41@USNAEXCH.na.didata.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Design Guide for Cisco Unity Virtualization
>
> <a href="http://www.cisco.com/en/US/docs/voice_ip_comm/unity/virtualization_desig">http://www.cisco.com/en/US/docs/voice_ip_comm/unity/virtualization_desig</a>
> n/guide/cuvirtualdgx.html
>
>
>
>
>
> Check the archives for this list, it was discussed last month -jason
>
>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Paul
> Sent: Wednesday, June 24, 2009 1:24 PM
> To: Miller, Steve; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on
> aPhysical Server and the Failover on a Virtual Server?
>
>
>
> First of all, is it supported?
>
> If it's not supported and you're planning on using it in a production
> environment, then yes you are crazy.
>
>
>
> ________________________________
>
> From: "Miller, Steve" <MillerS@DicksteinShapiro.COM>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Sent: Wednesday, June 24, 2009 1:20:31 PM
> Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a
> Physical Server and the Failover on a Virtual Server?
>
> We are looking to run Unity 7.0.2 on Server 2003. I am looking at a
> number of different scenarios, but I wanted to get a feel for whether
> this idea was totally crazy or just a little crazy.
>
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> <a href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</a>
>
>
>
> --------------------------------------------------------
>
> This e-mail message and any attached files are confidential and are
> intended solely for the use of the addressee(s)
>
> named above. This communication may contain material protected by
> attorney-client, work product, or other
>
> privileges. If you are not the intended recipient or person responsible
> for delivering this confidential
>
> communication to the intended recipient, you have received this
> communication in error, and any review, use,
>
> dissemination, forwarding, printing, copying, or other distribution of
> this e-mail message and any attached files
>
> is strictly prohibited. Dickstein Shapiro reserves the right to monitor
> any communication that is created,
>
> received, or sent on its network. If you have received this
> confidential communication in error, please notify the
>
> sender immediately by reply e-mail message and permanently delete the
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>
>
> To reply to our email administrator directly, send an email to
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>
>
>
> Dickstein Shapiro LLP
>
> <a href="http://www.DicksteinShapiro.com">http://www.DicksteinShapiro.com</a>
>
>
>
> ========================================================================
> ======
>
>
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> ------------------------------
>
> Message: 7
> Date: Wed, 24 Jun 2009 14:41:17 -0400
> From: Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>>
> To: Paul <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>>
> Cc: VOIP Group <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] FW: CME web access disable
> Message-ID:
> <<a href="mailto:56c3b48b0906241141s497c4e82l67cd5dbe7203c14c@mail.gmail.com">56c3b48b0906241141s497c4e82l67cd5dbe7203c14c@mail.gmail.com</a>>
> Content-Type: text/plain; charset=windows-1252
>
> telephony-service
> service phone webAccess 1
>
>
> Then reset the phone.
>
>
> Here's the full list of variables and their settings:
> <a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090">http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090</a>
>
>
> -nick
>
> On Wed, Jun 24, 2009 at 10:15 AM, Paul<<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>> wrote:
>
>
> Block port 80 on every switchport that has an IP phone on it.
>
>
>
>
> ________________________________
> From: Ahmed Elnagar <<a href="mailto:ahmed_elnagar@hotmail.com">ahmed_elnagar@hotmail.com</a>>
> To: VOIP Group <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Sent: Wednesday, June 24, 2009 3:38:19 AM
> Subject: [cisco-voip] FW: CME web access disable
>
>
>
> Hello all;
>
> Anyway know a way to disable phone web access for CME phones?
>
> ________________________________
> Windows Live?: Keep your life in sync. Check it out!
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
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>
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 24 Jun 2009 21:43:09 +0300
> From: Ahmed Elnagar <<a href="mailto:ahmed_elnagar@hotmail.com">ahmed_elnagar@hotmail.com</a>>
> To: <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>>, <<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>>
> Cc: VOIP Group <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] FW: CME web access disable
> Message-ID: <BLU106-W13192812FAE56DC190EA5C87370@phx.gbl>
> Content-Type: text/plain; charset="windows-1256"
>
>
>
> Nick...you are great thanks a lot really :)
>
> Thanks,
> Ahmed Elnagar
>
>
>
>
>
> Date: Wed, 24 Jun 2009 14:41:17 -0400
> Subject: Re: [cisco-voip] FW: CME web access disable
> From: <a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>
> To: <a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>
> CC: <a href="mailto:ahmed_elnagar@hotmail.com">ahmed_elnagar@hotmail.com</a>; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
>
> telephony-service
> service phone webAccess 1
>
>
> Then reset the phone.
>
>
> Here's the full list of variables and their settings:
> <a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090">http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090</a>
>
>
> -nick
>
> On Wed, Jun 24, 2009 at 10:15 AM, Paul<<a href="mailto:asobihoudai@yahoo.com">asobihoudai@yahoo.com</a>> wrote:
>
>
> Block port 80 on every switchport that has an IP phone on it.
>
>
>
>
> ________________________________
> From: Ahmed Elnagar <<a href="mailto:ahmed_elnagar@hotmail.com">ahmed_elnagar@hotmail.com</a>>
> To: VOIP Group <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Sent: Wednesday, June 24, 2009 3:38:19 AM
> Subject: [cisco-voip] FW: CME web access disable
>
>
>
> Hello all;
>
> Anyway know a way to disable phone web access for CME phones?
>
> ________________________________
> Windows Live?: Keep your life in sync. Check it out!
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
>
> _________________________________________________________________
> Show them the way! Add maps and directions to your party invites.
> <a href="http://www.microsoft.com/windows/windowslive/products/events.aspx">http://www.microsoft.com/windows/windowslive/products/events.aspx</a>
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>
> ------------------------------
>
> Message: 9
> Date: Wed, 24 Jun 2009 14:16:42 -0500
> From: "Jeff Ruttman" <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>
> To: "cisco-voip" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
> Message-ID:
> <<a href="mailto:07365C3161D8D8419EE51C3834C02205B84D47@ma1-exc01.ec2802.elderc.org">07365C3161D8D8419EE51C3834C02205B84D47@ma1-exc01.ec2802.elderc.org</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> Greetings,
>
> We're putting in a Verizon PRI at one of our offices. They're asking
> what Protocol we want, NAT1, NAT2, or custom. I believe this has to do
> with caller ID, and that "NAT" stands for National. Any insight into
> what I should choose?
>
> On our existing gateways with PRIs, the dropdowns in Call Routing
> Information where I could choose "National" we have chosen "Cisco Call
> Manager."
>
> Thanks
> jeff
>
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
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>
> ------------------------------
>
> Message: 10
> Date: Wed, 24 Jun 2009 15:22:41 -0400
> From: "Matt Slaga (US)" <<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>>
> To: Jeff Ruttman <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>, cisco-voip
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
> Message-ID:
> <5FE225375F6E3F4493474B90BC1B94EFB95160515B@USISPCLEXDB01.na.didata.local>
>
> Content-Type: text/plain; charset="us-ascii"
>
> You will want NI2, NI1 is not an option with Cisco gateways (surprised they are even willing to offer it, it's quite antiquated)..
>
> This is selected through the PRI protocol however, not through Call Routing information.
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jeff Ruttman
> Sent: Wednesday, June 24, 2009 3:17 PM
> To: cisco-voip
> Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
>
> Greetings,
>
> We're putting in a Verizon PRI at one of our offices. They're asking what Protocol we want, NAT1, NAT2, or custom. I believe this has to do with caller ID, and that "NAT" stands for National. Any insight into what I should choose?
>
> On our existing gateways with PRIs, the dropdowns in Call Routing Information where I could choose "National" we have chosen "Cisco Call Manager."
>
> Thanks
> jeff
>
>
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
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>
> ------------------------------
>
> Message: 11
> Date: Wed, 24 Jun 2009 14:30:55 -0500
> From: "Jeff Ruttman" <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>
> To: "Matt Slaga (US)" <<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>>, "cisco-voip"
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
> Message-ID:
> <<a href="mailto:07365C3161D8D8419EE51C3834C02205B84D49@ma1-exc01.ec2802.elderc.org">07365C3161D8D8419EE51C3834C02205B84D49@ma1-exc01.ec2802.elderc.org</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> Thanks Matt. Yes all our GWs with a PRI have NI2 as the PRI Protocol
> Type, and that was my first thought, but since NAT2 wasn't a option in
> the dropdown, I began looking elsewhere. So when Verizon says NAT2 that
> means NI2 in CCM?
>
> Thanks
> jeff
>
> ________________________________
>
> From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]
> Sent: Wednesday, June 24, 2009 2:23 PM
> To: Jeff Ruttman; cisco-voip
> Subject: RE: PRI Protocol NAT1, NAT2, custom?
>
>
>
> You will want NI2, NI1 is not an option with Cisco gateways (surprised
> they are even willing to offer it, it's quite antiquated)..
>
>
>
> This is selected through the PRI protocol however, not through Call
> Routing information.
>
>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jeff Ruttman
> Sent: Wednesday, June 24, 2009 3:17 PM
> To: cisco-voip
> Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
>
>
>
> Greetings,
>
>
>
> We're putting in a Verizon PRI at one of our offices. They're asking
> what Protocol we want, NAT1, NAT2, or custom. I believe this has to do
> with caller ID, and that "NAT" stands for National. Any insight into
> what I should choose?
>
>
>
> On our existing gateways with PRIs, the dropdowns in Call Routing
> Information where I could choose "National" we have chosen "Cisco Call
> Manager."
>
>
>
> Thanks
>
> jeff
>
>
>
>
>
>
>
> CONFIDENTIALITY NOTICE: The information contained in this email
> including attachments is intended for the specific delivery to and use
> by the individual(s) to whom it is addressed, and includes information
> which should be considered as private and confidential. Any review,
> retransmission, dissemination, or taking of any action in reliance upon
> this information by anyone other than the intended recipient is
> prohibited. If you have received this message in error, please reply to
> the sender immediately and delete the original message and any copy of
> it from your computer system. Thank you.
>
> ________________________________
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee,
> you are hereby notified that you have received this communication in
> error and that any use or reproduction of this email or its contents is
> strictly prohibited and may be unlawful. If you have received this
> communication in error, please notify us immediately by replying to this
> message and deleting it from your computer. Thank you.
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
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>
> ------------------------------
>
> Message: 12
> Date: Wed, 24 Jun 2009 15:34:57 -0400
> From: "Matt Slaga (US)" <<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>>
> To: Jeff Ruttman <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>, cisco-voip
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
> Message-ID:
> <5FE225375F6E3F4493474B90BC1B94EFB951605167@USISPCLEXDB01.na.didata.local>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Yes, NAT1 and NAT2 are short for National-1 and National-2. Cisco (and some others) call it NI which is short for National ISDN.
>
> So, you are right on track that you would select NI-2 for the telco's NAT-2.
>
> From: Jeff Ruttman [mailto:<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>]
> Sent: Wednesday, June 24, 2009 3:31 PM
> To: Matt Slaga (US); cisco-voip
> Subject: RE: PRI Protocol NAT1, NAT2, custom?
>
> Thanks Matt. Yes all our GWs with a PRI have NI2 as the PRI Protocol Type, and that was my first thought, but since NAT2 wasn't a option in the dropdown, I began looking elsewhere. So when Verizon says NAT2 that means NI2 in CCM?
>
> Thanks
> jeff
>
> ________________________________
> From: Matt Slaga (US) [mailto:<a href="mailto:Matt.Slaga@us.didata.com">Matt.Slaga@us.didata.com</a>]
> Sent: Wednesday, June 24, 2009 2:23 PM
> To: Jeff Ruttman; cisco-voip
> Subject: RE: PRI Protocol NAT1, NAT2, custom?
> You will want NI2, NI1 is not an option with Cisco gateways (surprised they are even willing to offer it, it's quite antiquated)..
>
> This is selected through the PRI protocol however, not through Call Routing information.
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jeff Ruttman
> Sent: Wednesday, June 24, 2009 3:17 PM
> To: cisco-voip
> Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
>
> Greetings,
>
> We're putting in a Verizon PRI at one of our offices. They're asking what Protocol we want, NAT1, NAT2, or custom. I believe this has to do with caller ID, and that "NAT" stands for National. Any insight into what I should choose?
>
> On our existing gateways with PRIs, the dropdowns in Call Routing Information where I could choose "National" we have chosen "Cisco Call Manager."
>
> Thanks
> jeff
>
>
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
> ________________________________
>
> Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
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>
> ------------------------------
>
> Message: 13
> Date: Wed, 24 Jun 2009 15:57:08 -0400
> From: Peter Slow <<a href="mailto:peter.slow@gmail.com">peter.slow@gmail.com</a>>
> To: Tim Frazee <<a href="mailto:tfrazee@gmail.com">tfrazee@gmail.com</a>>
> Cc: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] create a software CFB that got deleted
> Message-ID:
> <<a href="mailto:53fc16d40906241257r19946bc3qcd6282ad951f7dbd@mail.gmail.com">53fc16d40906241257r19946bc3qcd6282ad951f7dbd@mail.gmail.com</a>>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Try deactivating the IPVMS service, and then reactivating it. This is
> different from restarting the service. use service activation. Let us
> know how it goes.
>
> -Peter
>
> On Wed, Jun 24, 2009 at 1:43 PM, Tim Frazee<<a href="mailto:tfrazee@gmail.com">tfrazee@gmail.com</a>> wrote:
>
>
> hey all,
>
> Running CUCM 7.0(2a) and somehow my software CFB (the one that runs on the
> server, from the IPVM service) got deleted. How do I recreate the CFB on the
> server?
>
> Adding a software CFB is not a choice in the drop down menu when I try to
> just rebuild it.
>
> Any ideas?
>
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
>
>
>
> ------------------------------
>
> Message: 14
> Date: Wed, 24 Jun 2009 23:12:35 +0300
> From: Dew Swen <<a href="mailto:dew.swen@gmail.com">dew.swen@gmail.com</a>>
> To: Mehmet Turunc <<a href="mailto:turunc.mehmet@gmail.com">turunc.mehmet@gmail.com</a>>
> Cc: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] destination-pattern "T" question
> Message-ID:
> <<a href="mailto:ae5778960906241312j5eb22bf5g91fcc7abceea99d5@mail.gmail.com">ae5778960906241312j5eb22bf5g91fcc7abceea99d5@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Well, let me tell u.
>
> Matching occurs digit by digit unless en-bloc is not been configured.
>
> The number is "90114989123456"
>
> When it is press to 9, none of the dial peers are matched.
>
> After 0 is pressed dial-peer 90 is matched beacuse of T parameter which
> collects all digits. However, dial-peer 90110 still does not match.
>
> If dial-peer 90 does not exist, dial-peer 90110 matches "after all the 9011
> digits are pressed, and another digit is pressed".
>
>
> On the other hand, if en-bloc is enabled, all digits are sent at the same
> time. So 9T and 9011T are being processed at the same time. Because being a
> longer prefix, dial-peer 90110 matches.
>
> Hope it is clear.
>
> Regards,
> *
> -
> Dew Swen*
>
>
> On Tue, Jun 23, 2009 at 12:44 PM, Mehmet Turunc <<a href="mailto:turunc.mehmet@gmail.com">turunc.mehmet@gmail.com</a>>wrote:
>
>
>
> Hi all,
>
> I was studying Cisco Voice over IP (CVOICE) -Kevin Wallace 2009- and didn't
> understand this example, so I'm confused. Probably a newbee issue:)
>
> Router(config)#dial-peer voice 90 pots
> Router(config-dial-peer)#destination-pattern 9T
> Router(config-dial-peer)#port 0/0/0:23
> Router(config-dial-peer)#exit
> Router(config)#dial-peer voice 90110 pots
> Router(config-dial-peer)#destination-pattern 9011T
> Router(config-dial-peer)#port 0/0/1:23
>
> And the explanation:
>
> The following steps describe what occurs during the call in this example.
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because the first digit received is a 9, the gateway performs dial-peer
> matching.
> 3. Dial-peer 90 is matched, and any further digits are collected by the
> control character
> T that indicates the destination-pattern value is a variable-length dial
> string. (WHY? why doesnt longest prefix match?)
> 4. The user finishes dialing, and the call is routed using dial-peer 90.
> Dial-peer 90110
> will never be considered.
>
>
> For en bloc signaling, the DNIS is used, so the process is as follows:
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because en bloc signaling is enabled, the gateway continues to collect
> digits until the
> interdigit timeout value is exceeded.
> 3. The user finishes dialing, and the call is routed using dial-peer 90110.
>
> Thanks for the help
>
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
>
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>
> ------------------------------
>
> Message: 15
> Date: Wed, 24 Jun 2009 16:44:39 -0400
> From: "Jason Aarons (US)" <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>
> To: <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
> 2833 for DTMF
> Message-ID:
> <C1FE15183DA37645BC0633BC604E44F00F48FCF1@USNAEXCH.na.didata.local>
> Content-Type: text/plain; charset="us-ascii"
>
> I have a SIP Trunk from Verizon Business running thru ACME Packet box to
> CallManager 7.1(2a) which then routes to users voicemail on Unity
> Connection 7.1.2a connected via SIP trunk.
>
>
>
> If I press Zero or another dtmf key press does Unity support RFC2833 for
> DTMF or is a dynamic MPT resource needing to be invoked ?
>
>
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
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>
> ------------------------------
>
> Message: 16
> Date: Wed, 24 Jun 2009 12:49:31 -0700
> From: Tony Underwood <<a href="mailto:tony@cambiumdata.com">tony@cambiumdata.com</a>>
> To: Daniel <<a href="mailto:dan.voip@danofive.id.au">dan.voip@danofive.id.au</a>>, "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>"
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Multicast MoH Delay
> Message-ID:
> <<a href="mailto:0F205F18DCB4724DB15EAF8FF93E0A21129EAEE6FE@P3PW5EX1MB04.EX1.SECURESERVER.NET">0F205F18DCB4724DB15EAF8FF93E0A21129EAEE6FE@P3PW5EX1MB04.EX1.SECURESERVER.NET</a>>
>
> Content-Type: text/plain; charset="us-ascii"
>
> If it's a delay in the route set up then you could try a static igmp join on the far end router.
> ip igmp join-group group-address
>
> Tony Underwood CCIE #7112
> Sr. Network Engineer
> Cambium Data Inc.
> 5050 So. 111th St.
> Omaha, NE 68137
> (402) 556-1388
> <a href="http://www.cambiumdata.com">http://www.cambiumdata.com</a><<a href="http://www.cambiumdata.com/">http://www.cambiumdata.com/</a>>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Daniel
> Sent: Wednesday, June 24, 2009 1:46 AM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] Multicast MoH Delay
>
> Hi All,
>
> I've setup multicast routing for music on hold between our data centre network and building floor subnets. The setup is that the data centre is on a different subnet and gear to the floor subnets so multicast routing needs to be used to get MoH packets to the phones.
>
> This consists of the following traffic flow,
>
> MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution Switch/router > Floor Switches > Phones
>
> There are three routing hops (1) from vlan interface to routed interfaces of distribution switch (2) between Distribution and (3) from routed interface of distributionswitch to phone vlan interface. Only the distribution switches are layer 3.
>
> The first distribution switch is the RP for the group and only that group.The second distribution switch is accepting auto RP only. We are using sparse mode.
>
> Multicast traffic is working, the RP mappings are there, the mroute is there, the problem is that when a call is placed on hold there is a 5 second delay before the music is heard. I assume this to be somewhat because of the join message and the delay of the route or path being setup.
>
> Anyone know of a way to reduce the delay before music is heard? If not I guess its back to the lab.
>
> regards,
>
> Dan
>
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 17
> Date: Thu, 25 Jun 2009 08:44:12 +1000
> From: Daniel <<a href="mailto:dan.voip@danofive.id.au">dan.voip@danofive.id.au</a>>
> To: "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Multicast MoH Delay
> Message-ID:
> <<a href="mailto:f861d63a0906241544t965b6c6x53696a9ddd691e7f@mail.gmail.com">f861d63a0906241544t965b6c6x53696a9ddd691e7f@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanks all for your replies
>
> I would rather use sparse mode to take use of the join messages etc.. so
> that the phones join and leave the group. Dense mode I think floods the
> network and then prunes interfaces that are not in use, this occurs every 3
> minuts or so. It just seems that sparse mode is more precise, I don't have
> much experience in this so not sure. Any thoughts / technical reasons on why
> to go Sparse, Dense or both?
>
> The RP is on a 6500 with sup720s and MSFC3's, the other distribution switch
> is an older 6500 with SUP2s and MSFC2, so c3voip similair to the issue you
> had with TAC but the RP is on a different device.
>
> I have added the "ip igmp join-group group-address" command in, this fixes
> the issue. My question is with this command, the router will accept and
> forward these packets preventing fast switching, if i use the static-group
> command the router doesnt accept the packets itself but forwards them thus
> allowing fast switching, anyone know of benefits either way? I did
> originally have a look at this command but I assumed its use was to always
> have the multicast traffic flowing which I didnt think ideal. But it turns
> out after actually trying this from Tony's point below it works quite well.
> Between the dsitribution switches the mroute is always setup but not from
> the distribution switch to the floors, when the phone joins the group the
> phones vlan interface is added to the mroute and MoH is heard straight away.
> There is no multicast MoH flooding the floor vlans which is what I was
> concerned about.
>
>
>
>
>
>
>
>
> On Thu, Jun 25, 2009 at 5:49 AM, Tony Underwood <<a href="mailto:tony@cambiumdata.com">tony@cambiumdata.com</a>>wrote:
>
>
>
> If it's a delay in the route set up then you could try a static igmp join
> on the far end router.
>
> *ip igmp join-group **group-address*
>
>
>
> *Tony Underwood CCIE #7112*
>
> Sr. Network Engineer
>
> Cambium Data Inc.
>
> 5050 So. 111th St.
>
> Omaha, NE 68137
>
> (402) 556-1388
>
> <a href="http://www.cambiumdata.com">http://www.cambiumdata.com</a>
>
>
>
> *From:* <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:
> <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] *On Behalf Of *Daniel
> *Sent:* Wednesday, June 24, 2009 1:46 AM
> *To:* <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> *Subject:* [cisco-voip] Multicast MoH Delay
>
>
>
> Hi All,
>
>
>
> I've setup multicast routing for music on hold between our data centre
> network and building floor subnets. The setup is that the data centre is on
> a different subnet and gear to the floor subnets so multicast routing needs
> to be used to get MoH packets to the phones.
>
>
>
> This consists of the following traffic flow,
>
>
>
> MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution
> Switch/router > Floor Switches > Phones
>
>
>
> There are three routing hops (1) from vlan interface to routed interfaces
> of distribution switch (2) between Distribution and (3) from routed
> interface of distributionswitch to phone vlan interface. Only the
> distribution switches are layer 3.
>
>
>
> The first distribution switch is the RP for the group and only that
> group.The second distribution switch is accepting auto RP only. We are using
> sparse mode.
>
>
>
> Multicast traffic is working, the RP mappings are there, the mroute is
> there, the problem is that when a call is placed on hold there is a 5 second
> delay before the music is heard. I assume this to be somewhat because of the
> join message and the delay of the route or path being setup.
>
>
>
> Anyone know of a way to reduce the delay before music is heard? If not I
> guess its back to the lab.
>
>
>
> regards,
>
>
>
> Dan
>
>
>
>
>
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 18
> Date: Wed, 24 Jun 2009 15:00:37 -0700 (PDT)
> From: Jake Doe <<a href="mailto:jd80301@yahoo.com">jd80301@yahoo.com</a>>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] SIP Route Pattern
> Message-ID: <<a href="mailto:873114.16995.qm@web50806.mail.re2.yahoo.com">873114.16995.qm@web50806.mail.re2.yahoo.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello.
>
> We just upgraded to CUCM 7.1.2.20000-2 and are trying to add a SIP Route Pattern.? However, we are getting the following error:
>
> Add failed. [25256] International Strip Digits should be empty for devices other than H323 gateways and trunks and MGCP T1/E1 PRI and BRI gateways
>
> Any ideas how to correct this problem?? Also, I just noticed that we are using demo licenses.? Could this be causing the issue above?
>
> Thanks.
>
> JD
>
>
>
>
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 19
> Date: Wed, 24 Jun 2009 19:19:13 -0400
> From: "Jason Aarons (US)" <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>
> To: <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] TAPS and UCCX 5.0.2 ?
> Message-ID:
> <C1FE15183DA37645BC0633BC604E44F00F4D14F7@USNAEXCH.na.didata.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Customer is using CallManager 7.1 with off-box CRS 5.0.2 MCS-7845 server
> for TAPS (Tool for Auto-Registered Phones Support). Currently I can have
> up to 5 phones connecting running the TAPS .aef script.
>
> What is the UCCX license part number to increase the number of ports for
> TAPS?
>
> They currently have 150IVR ports and I assume 5 Agent Licenses? Does
> TAPS use Agent Licenses?
>
> Or I suspect I don't need Agent licenses and that in AppAdmin on UCCX
> under Trigger and/or Media Termination Dialog Group they might currently
> be set to 5 and just need to be increased to 150 to allow 150 sessions
> of TAPS?
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
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>
> ------------------------------
>
> Message: 20
> Date: Wed, 24 Jun 2009 20:34:25 -0400
> From: "Adam Frankel" <<a href="mailto:afrankel@cisco.com">afrankel@cisco.com</a>>
> To: "'Jason Aarons \(US\)'" <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>,
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
> 2833 for DTMF
> Message-ID: <007201c9f52c$b1453a00$13cfae00$@com>
> Content-Type: text/plain; charset="us-ascii"
>
> Jason,
>
>
>
> I believe it does (don't quote me on that) but one way to tell would be to
> check the CCM traces for the Capabilities Response sent by the Unity port
> when it registers with CUCM. Check for 257.
>
>
>
> Adam
>
>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jason Aarons (US)
> Sent: Wednesday, June 24, 2009 4:45 PM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC 2833 for
> DTMF
>
>
>
> I have a SIP Trunk from Verizon Business running thru ACME Packet box to
> CallManager 7.1(2a) which then routes to users voicemail on Unity Connection
> 7.1.2a connected via SIP trunk.
>
>
>
> If I press Zero or another dtmf key press does Unity support RFC2833 for
> DTMF or is a dynamic MPT resource needing to be invoked ?
>
>
>
> _____
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you.
>
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>
> ------------------------------
>
> Message: 21
> Date: Wed, 24 Jun 2009 19:45:42 -0500
> From: Erick Bergquist <<a href="mailto:erickbee@gmail.com">erickbee@gmail.com</a>>
> To: Jason Burns <<a href="mailto:burns.jason@gmail.com">burns.jason@gmail.com</a>>
> Cc: cisco-voip mailinglist <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Call Manager 5.1.1a install file?
> Message-ID:
> <<a href="mailto:f4445faf0906241745q7b616ea6p3c44f01d8374fd6@mail.gmail.com">f4445faf0906241745q7b616ea6p3c44f01d8374fd6@mail.gmail.com</a>>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Yea, thanks.
>
> On Wed, Jun 24, 2009 at 1:28 PM, Jason Burns<<a href="mailto:burns.jason@gmail.com">burns.jason@gmail.com</a>> wrote:
>
>
> That would probably be 5.1.1.2000-1 or 2.
>
> On Wed, Jun 24, 2009 at 2:06 PM, Erick Bergquist <<a href="mailto:erickbee@gmail.com">erickbee@gmail.com</a>> wrote:
>
>
> Does anyone recall what the install file / version is for CUCM 5.1.1a?
> ?is it 3000-4?
>
> Thanks, Erick
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
>
>
>
> ------------------------------
>
> Message: 22
> Date: Thu, 25 Jun 2009 11:19:57 +1000
> From: ciscozest <<a href="mailto:ciscozest@gmail.com">ciscozest@gmail.com</a>>
> To: cisco-voip mailinglist <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] T.37 Fax Redundancy
> Message-ID:
> <<a href="mailto:f99cc3f60906241819l32750916n541060c9184d049e@mail.gmail.com">f99cc3f60906241819l32750916n541060c9184d049e@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> we are planning for on-ramp T.37 store and forward fax and wondering about
> the redundancy of this way. Can anyone enlighten me on this? We have an
> on-ramp T.37 gateway at site A while the IP fax server is located in
> different site. Connectivity is over the WAN.
>
> 1. What happen to the active fax session when the WAN link is down? Would
> the gateway keep trying to reach the Fax server few times and then give up
> and drop the fax content?
> 2. What happen to the NEW incoming fax session when the WAN link is down?
> Would it be stored locally in IOS gateway which run the T.37 protocol?
> 3. Can I create another dial-peer for the T.37 fax number with higher
> preference value and push it to the local fax mahine attached to the FXS
> port on the on-ramp gateway?
>
> Thank you
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 23
> Date: Wed, 24 Jun 2009 18:42:42 -0700
> From: Tony Underwood <<a href="mailto:tony@cambiumdata.com">tony@cambiumdata.com</a>>
> To: Daniel <<a href="mailto:dan.voip@danofive.id.au">dan.voip@danofive.id.au</a>>, "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>"
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Multicast MoH Delay
> Message-ID:
> <<a href="mailto:0F205F18DCB4724DB15EAF8FF93E0A21129EAEE76A@P3PW5EX1MB04.EX1.SECURESERVER.NET">0F205F18DCB4724DB15EAF8FF93E0A21129EAEE76A@P3PW5EX1MB04.EX1.SECURESERVER.NET</a>>
>
> Content-Type: text/plain; charset="us-ascii"
>
> I'm the furthest thing from a Multicast expert, but to my knowledge the static join is joining only the router interface to the multicast stream and it forwards the packets continuously. But, when the packets get to the L2 switch, it doesn't have an active IGMP join in it's table so it doesn't forward the traffic out any ports. Then when the phone joins the group it is instantly available at the access layer due to the static join.
> So, if anything this tells you that your delay is between the L3 devices and not at the access layer.
>
> Tony Underwood CCIE #7112
> Sr. Network Engineer
> Cambium Data Inc.
> 5050 So. 111th St.
> Omaha, NE 68137
> (402) 556-1388
> <a href="http://www.cambiumdata.com">http://www.cambiumdata.com</a><<a href="http://www.cambiumdata.com/">http://www.cambiumdata.com/</a>>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Daniel
> Sent: Wednesday, June 24, 2009 5:44 PM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] Multicast MoH Delay
>
> Thanks all for your replies
>
> I would rather use sparse mode to take use of the join messages etc.. so that the phones join and leave the group. Dense mode I think floods the network and then prunes interfaces that are not in use, this occurs every 3 minuts or so. It just seems that sparse mode is more precise, I don't have much experience in this so not sure. Any thoughts / technical reasons on why to go Sparse, Dense or both?
>
> The RP is on a 6500 with sup720s and MSFC3's, the other distribution switch is an older 6500 with SUP2s and MSFC2, so c3voip similair to the issue you had with TAC but the RP is on a different device.
>
> I have added the "ip igmp join-group group-address" command in, this fixes the issue. My question is with this command, the router will accept and forward these packets preventing fast switching, if i use the static-group command the router doesnt accept the packets itself but forwards them thus allowing fast switching, anyone know of benefits either way? I did originally have a look at this command but I assumed its use was to always have the multicast traffic flowing which I didnt think ideal. But it turns out after actually trying this from Tony's point below it works quite well. Between the dsitribution switches the mroute is always setup but not from the distribution switch to the floors, when the phone joins the group the phones vlan interface is added to the mroute and MoH is heard straight away. There is no multicast MoH flooding the floor vlans which is what I was concerned about.
>
>
>
>
>
>
>
>
> On Thu, Jun 25, 2009 at 5:49 AM, Tony Underwood <<a href="mailto:tony@cambiumdata.com">tony@cambiumdata.com</a><mailto:<a href="mailto:tony@cambiumdata.com">tony@cambiumdata.com</a>>> wrote:
>
> If it's a delay in the route set up then you could try a static igmp join on the far end router.
>
> ip igmp join-group group-address
>
>
>
> Tony Underwood CCIE #7112
>
> Sr. Network Engineer
>
> Cambium Data Inc.
>
> 5050 So. 111th St.
>
> Omaha, NE 68137
>
> (402) 556-1388
>
> <a href="http://www.cambiumdata.com">http://www.cambiumdata.com</a><<a href="http://www.cambiumdata.com/">http://www.cambiumdata.com/</a>>
>
>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>>] On Behalf Of Daniel
> Sent: Wednesday, June 24, 2009 1:46 AM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><mailto:<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] Multicast MoH Delay
>
>
>
> Hi All,
>
>
>
> I've setup multicast routing for music on hold between our data centre network and building floor subnets. The setup is that the data centre is on a different subnet and gear to the floor subnets so multicast routing needs to be used to get MoH packets to the phones.
>
>
>
> This consists of the following traffic flow,
>
>
>
> MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution Switch/router > Floor Switches > Phones
>
>
>
> There are three routing hops (1) from vlan interface to routed interfaces of distribution switch (2) between Distribution and (3) from routed interface of distributionswitch to phone vlan interface. Only the distribution switches are layer 3.
>
>
>
> The first distribution switch is the RP for the group and only that group.The second distribution switch is accepting auto RP only. We are using sparse mode.
>
>
>
> Multicast traffic is working, the RP mappings are there, the mroute is there, the problem is that when a call is placed on hold there is a 5 second delay before the music is heard. I assume this to be somewhat because of the join message and the delay of the route or path being setup.
>
>
>
> Anyone know of a way to reduce the delay before music is heard? If not I guess its back to the lab.
>
>
>
> regards,
>
>
>
> Dan
>
>
>
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>
> ------------------------------
>
> Message: 24
> Date: Wed, 24 Jun 2009 22:56:15 -0400
> From: "Miller, Steve" <MillerS@DicksteinShapiro.COM>
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] One User Cannot Be Dialed By Name
> Message-ID: <<a href="mailto:418329B7ED67E64BBD2BAA97078D70D001C62118@DCEX2.DSMO.COM">418329B7ED67E64BBD2BAA97078D70D001C62118@DCEX2.DSMO.COM</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> Any ideas why this happens? This one person cannot be dialed by name.
> I've checked his name and tried to do it myself, but cannot.
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> <a href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</a>
>
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
> named above. This communication may contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for delivering this confidential
> communication to the intended recipient, you have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
> is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
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> sender immediately by reply e-mail message and permanently delete the original message.
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> To reply to our email administrator directly, send an email to <a href="mailto:postmaster@dicksteinshapiro.com">postmaster@dicksteinshapiro.com</a>
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> Dickstein Shapiro LLP
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>
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> ------------------------------
>
> Message: 25
> Date: Wed, 24 Jun 2009 20:00:39 -0700
> From: Cristobal Priego <<a href="mailto:cristobalpriego@gmail.com">cristobalpriego@gmail.com</a>>
> To: "Miller, Steve" <MillerS@DicksteinShapiro.COM>
> Cc: "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] One User Cannot Be Dialed By Name
> Message-ID: <<a href="mailto:3E4765E5-CCD9-485C-8FC9-949F63DB0E93@gmail.com">3E4765E5-CCD9-485C-8FC9-949F63DB0E93@gmail.com</a>>
> Content-Type: text/plain; charset="us-ascii"; Format="flowed";
> DelSp="yes"
>
> Please make sure that the user has a recorded name and that the option
> for list in directory is checked
>
> Sent from my iPhone
>
> On Jun 24, 2009, at 7:56 PM, "Miller, Steve" <MillerS@DicksteinShapiro.COM
> > wrote:
>
>
>
> Any ideas why this happens? This one person cannot be dialed by
> name. I've checked his name and tried to do it myself, but cannot.
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> <a href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</a>
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are
> intended solely for the use of the addressee(s)
> named above. This communication may contain material protected by
> attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential
> communication to the intended recipient, you have received this
> communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution
> of this e-mail message and any attached files
> is strictly prohibited. Dickstein Shapiro reserves the right to
> monitor any communication that is created,
> received, or sent on its network. If you have received this
> confidential communication in error, please notify the
> sender immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email to <a href="mailto:postmaster@dicksteinshapiro.com">postmaster@dicksteinshapiro.com</a>
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> Dickstein Shapiro LLP
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>
> ===
> ===
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> =====================================================================
> _______________________________________________
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>
> ------------------------------
>
> Message: 26
> Date: Wed, 24 Jun 2009 23:44:08 -0400
> From: Dustin S Fowler <<a href="mailto:dustin.s.fowler@gmail.com">dustin.s.fowler@gmail.com</a>>
> To: "Jason Aarons (US)" <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>
> Cc: "<a href="mailto:cisco-voip@puck-nether.net">cisco-voip@puck-nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] TAPS and UCCX 5.0.2 ?
> Message-ID:
> <<a href="mailto:f2d16ce0906242044r1ce0dc65jd1eb8c3d31375e05@mail.gmail.com">f2d16ce0906242044r1ce0dc65jd1eb8c3d31375e05@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Jason,
>
> Your setting would need to be changed in UCCX App admin. There are a few
> spots but you should look for the 'sessions'. I would starts at the trigger
> and work your way to the Media Termination Group.
>
> Let us know how it goes.
>
> Dustin Fowler
>
>
>
> On Wed, Jun 24, 2009 at 7:19 PM, Jason Aarons (US) <
> <a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>> wrote:
>
>
>
> Customer is using CallManager 7.1 with off-box CRS 5.0.2 MCS-7845 server
> for TAPS (Tool for Auto-Registered Phones Support). Currently I can have up
> to 5 phones connecting running the TAPS .aef script.
>
> What is the UCCX license part number to increase the number of ports for
> TAPS?
>
> They currently have 150IVR ports and I assume 5 Agent Licenses? Does TAPS
> use Agent Licenses?
>
> Or I suspect I don't need Agent licenses and that in AppAdmin on UCCX under
> Trigger and/or Media Termination Dialog Group they might currently be set to
> 5 and just need to be increased to 150 to allow 150 sessions of TAPS?
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
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> _______________________________________________
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>
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>
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>
> ------------------------------
>
> Message: 27
> Date: Wed, 24 Jun 2009 21:07:31 -0700
> From: Mark Holloway <<a href="mailto:mh@markholloway.com">mh@markholloway.com</a>>
> To: Jason Aarons (US) <<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>>
> Cc: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
> 2833 for DTMF
> Message-ID: <<a href="mailto:F6250C96-D4DD-4FAD-BDCC-E64DF7B9C442@markholloway.com">F6250C96-D4DD-4FAD-BDCC-E64DF7B9C442@markholloway.com</a>>
> Content-Type: text/plain; charset="us-ascii"; Format="flowed";
> DelSp="yes"
>
> Unity Connections should already support RFC 2833. The Acme Session
> Director realms can be configured for Transparent DTMF (RFC 2833 to
> RFC 2833, or SIP Info to SIP Info), RFC 2833 to SIP Info, SIP Info to
> RFC 2833, or dual-mode where one realm is set to Transparent and the
> other realm (facing UC) sends both event types. This is common where
> one SIP server requires one DTMF type then redirects the call to
> another SIP server that uses another DTMF type.
>
>
>
> On Jun 24, 2009, at 1:44 PM, Jason Aarons (US) wrote:
>
>
>
> I have a SIP Trunk from Verizon Business running thru ACME Packet
> box to CallManager 7.1(2a) which then routes to users voicemail on
> Unity Connection 7.1.2a connected via SIP trunk.
>
> If I press Zero or another dtmf key press does Unity support RFC2833
> for DTMF or is a dynamic MPT resource needing to be invoked ?
>
>
>
> Disclaimer: This e-mail communication and any attachments may
> contain confidential and privileged information and is for use by
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> _______________________________________________
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>
> ------------------------------
>
> Message: 28
> Date: Thu, 25 Jun 2009 02:59:33 -0300
> From: "ROJAS, Mario" <<a href="mailto:Mario.ROJAS@LA.LOGICALIS.COM">Mario.ROJAS@LA.LOGICALIS.COM</a>>
> To: <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] Show Saved Enterprise Data in CAD
> Message-ID:
> <<a href="mailto:1081D526E13AB442851D3AE541A2F112CBBB8C@SNLAR-EXCH01.LA.LOGICALIS.COM">1081D526E13AB442851D3AE541A2F112CBBB8C@SNLAR-EXCH01.LA.LOGICALIS.COM</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Hello,
>
> We are working on a proposal of a Unified CCX 7, and the customer wants the following to happen:
>
> 1) The agent answers a call and has fields in his desktop client to classify the call (like, New Customer, VIP, etc). I know this can be done with customizable Enterprise Data.
>
> 2) The next time the agent answers the call coming from the same telephone number (or whatever method of identifying the called, like a customer ID), the agent desktop shows the last variable saved. Like, in step 1, the first call was classified as a New Customer. The next time the a call from the same number goes in, the agent can see how the previous call was treated.
>
> Is that possible? I have configured custom Enterprise Data fields, and I can save information on them, but they don't show up the next time the call comes in.
>
> Best regards,
>
> MARIO ROJAS GUERRERO
> Systems Engineer
>
>
> LOGICALIS
> Los Sauces 325 - San Isidro
>
> Lima 27 - Per?
> Tel/Fax: +51-1 611-9682
> Mov:+51-1 980300124
> <mailto:<a href="mailto:mario.rojas@la.logicalis.com">mario.rojas@la.logicalis.com</a>> <a href="mailto:mario.rojas@la.logicalis.com">mario.rojas@la.logicalis.com</a>
> <<a href="http://www.la.logicalis.com">http://www.la.logicalis.com</a>> <a href="http://www.la.logicalis.com">www.la.logicalis.com</a>
> <<a href="http://www.logicalisnow.com/">http://www.logicalisnow.com/</a>> <a href="http://www.logicalisnow.com">www.logicalisnow.com</a>
>
>
> Por favor, piense en el medioambiente antes de imprimir este email.
> La presente informaci?n se env?a ?nicamente para el destinatario, y contiene informaci?n de car?cter CONFIDENCIAL o PRIVLEGIADA.
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>
> Please, think about the environment before printing this email.
>
> The present information is sent solely for the adressee, and contains information of CONFIDENTIAL or PRIVILEGED nature. The modification, broadcasting, diffusion, copy or another use of this information by any means, of people different from the adressee, are strictly prohibited.
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>
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>
> ------------------------------
>
> Message: 29
> Date: Thu, 25 Jun 2009 11:17:58 +0300
> From: Mehmet Turunc <<a href="mailto:turunc.mehmet@gmail.com">turunc.mehmet@gmail.com</a>>
> To: Dew Swen <<a href="mailto:dew.swen@gmail.com">dew.swen@gmail.com</a>>
> Cc: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: Re: [cisco-voip] destination-pattern "T" question
> Message-ID:
> <<a href="mailto:7d505d120906250117n274614c7j16d6148da9a1102b@mail.gmail.com">7d505d120906250117n274614c7j16d6148da9a1102b@mail.gmail.com</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanx for the reply Dew. I understand the general idea of your response. But
> I couldn't understand some points.
>
> When I open "debug dial-peer voice" debugging, and after that starting to
> dial digits, digit by digit matching happens. So, does it mean "by default
> digit by digit analysis happens"?
>
> For enabling en-bloc signaling which command should i use? I couldn't find
> more specific details.
>
>
>
> On Wed, Jun 24, 2009 at 11:12 PM, Dew Swen <<a href="mailto:dew.swen@gmail.com">dew.swen@gmail.com</a>> wrote:
>
>
>
> Well, let me tell u.
>
> Matching occurs digit by digit unless en-bloc is not been configured.
>
> The number is "90114989123456"
>
> When it is press to 9, none of the dial peers are matched.
>
> After 0 is pressed dial-peer 90 is matched beacuse of T parameter which
> collects all digits. However, dial-peer 90110 still does not match.
>
> If dial-peer 90 does not exist, dial-peer 90110 matches "after all the 9011
> digits are pressed, and another digit is pressed".
>
>
> On the other hand, if en-bloc is enabled, all digits are sent at the same
> time. So 9T and 9011T are being processed at the same time. Because being a
> longer prefix, dial-peer 90110 matches.
>
> Hope it is clear.
>
> Regards,
> *
> -
> Dew Swen*
>
>
> On Tue, Jun 23, 2009 at 12:44 PM, Mehmet Turunc <<a href="mailto:turunc.mehmet@gmail.com">turunc.mehmet@gmail.com</a>>wrote:
>
>
>
> Hi all,
>
> I was studying Cisco Voice over IP (CVOICE) -Kevin Wallace 2009- and
> didn't understand this example, so I'm confused. Probably a newbee issue:)
>
> Router(config)#dial-peer voice 90 pots
> Router(config-dial-peer)#destination-pattern 9T
> Router(config-dial-peer)#port 0/0/0:23
> Router(config-dial-peer)#exit
> Router(config)#dial-peer voice 90110 pots
> Router(config-dial-peer)#destination-pattern 9011T
> Router(config-dial-peer)#port 0/0/1:23
>
> And the explanation:
>
> The following steps describe what occurs during the call in this example.
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because the first digit received is a 9, the gateway performs dial-peer
> matching.
> 3. Dial-peer 90 is matched, and any further digits are collected by the
> control character
> T that indicates the destination-pattern value is a variable-length dial
> string. (WHY? why doesnt longest prefix match?)
> 4. The user finishes dialing, and the call is routed using dial-peer 90.
> Dial-peer 90110
> will never be considered.
>
>
> For en bloc signaling, the DNIS is used, so the process is as follows:
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because en bloc signaling is enabled, the gateway continues to collect
> digits until the
> interdigit timeout value is exceeded.
> 3. The user finishes dialing, and the call is routed using dial-peer
> 90110.
>
> Thanks for the help
>
> _______________________________________________
> cisco-voip mailing list
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a>
>
>
>
>
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>
> ------------------------------
>
> Message: 30
> Date: Thu, 25 Jun 2009 08:37:29 -0500
> From: "Beck, Christopher" <<a href="mailto:CBeck@usg.com">CBeck@usg.com</a>>
> To: "ROJAS, Mario" <<a href="mailto:Mario.ROJAS@LA.LOGICALIS.COM">Mario.ROJAS@LA.LOGICALIS.COM</a>>,
> "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Show Saved Enterprise Data in CAD
> Message-ID:
> <<a href="mailto:AF8E5A5BA2DF074E8B46A3B4A8BC265513806156@CERO-MB-01.USG.NET">AF8E5A5BA2DF074E8B46A3B4A8BC265513806156@CERO-MB-01.USG.NET</a>>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Others may correct me for this, but I believe you are going to need to integrate a database into this mix to store those variables. Thus, you can use ANI to do a lookup and read these values into your variables prior to presenting the call to an agent.
>
> Chris Beck
> IT Lead - Voice Technologies
> USG Corporation
> 312-436-4541 (office)
> 312-730-5524 (Mobile)
> 312-672-4541 (FAX)
> <a href="mailto:cbeck@usg.com">cbeck@usg.com</a>
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of ROJAS, Mario
> Sent: Thursday, June 25, 2009 1:00 AM
> To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>
> Subject: [cisco-voip] Show Saved Enterprise Data in CAD
>
> Hello,
>
> We are working on a proposal of a Unified CCX 7, and the customer wants the following to happen:
>
> 1) The agent answers a call and has fields in his desktop client to classify the call (like, New Customer, VIP, etc). I know this can be done with customizable Enterprise Data.
>
> 2) The next time the agent answers the call coming from the same telephone number (or whatever method of identifying the called, like a customer ID), the agent desktop shows the last variable saved. Like, in step 1, the first call was classified as a New Customer. The next time the a call from the same number goes in, the agent can see how the previous call was treated.
>
> Is that possible? I have configured custom Enterprise Data fields, and I can save information on them, but they don't show up the next time the call comes in.
>
> Best regards,
>
> MARIO ROJAS GUERRERO
> Systems Engineer
>
> LOGICALIS
> Los Sauces 325 - San Isidro
> Lima 27 - Per?
> Tel/Fax: +51-1 611-9682
> Mov:+51-1 980300124
> <a href="mailto:mario.rojas@la.logicalis.com">mario.rojas@la.logicalis.com</a><mailto:<a href="mailto:mario.rojas@la.logicalis.com">mario.rojas@la.logicalis.com</a>>
> <a href="http://www.la.logicalis.com">www.la.logicalis.com</a><<a href="http://www.la.logicalis.com">http://www.la.logicalis.com</a>>
> <a href="http://www.logicalisnow.com">www.logicalisnow.com</a><<a href="http://www.logicalisnow.com/">http://www.logicalisnow.com/</a>>
>
> Por favor, piense en el medioambiente antes de imprimir este email.
> La presente informaci?n se env?a ?nicamente para el destinatario, y contiene informaci?n de car?cter CONFIDENCIAL o PRIVLEGIADA.
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> Please, think about the environment before printing this email.
> The present information is sent solely for the adressee, and contains information of CONFIDENTIAL or PRIVILEGED nature. The modification, broadcasting, diffusion, copy or another use of this information by any means, of people different from the adressee, are strictly prohibited.
>
>
>
>
> __________ Information from ESET Smart Security, version of virus signature database 4187 (20090625) __________
>
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> <a href="http://www.eset.com">http://www.eset.com</a>
>
>
> Confidentiality Notice: This email is intended for the sole use of the intended recipient(s) and may contain confidential, proprietary or privileged information. If you are not the intended recipient, you are notified that any use, review, dissemination, copying or action taken based on this message or its attachments, if any, is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy or delete all copies of the original message and any attachments. Thank you.
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>
> ------------------------------
>
> Message: 31
> Date: Thu, 25 Jun 2009 08:47:04 -0500
> From: "Jeff Ruttman" <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>
> To: "cisco-voip" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] Slow to connect calls
> Message-ID:
> <<a href="mailto:07365C3161D8D8419EE51C3834C02205B84D50@ma1-exc01.ec2802.elderc.org">07365C3161D8D8419EE51C3834C02205B84D50@ma1-exc01.ec2802.elderc.org</a>>
> Content-Type: text/plain; charset="us-ascii"
>
> Greetings,
>
> Some of our sites have DID trunk ports and POTS lines, and we have MGCP
> controlled GWs with FXS and FXO configured. We also have for these
> sites H.323 GWs--which frankly I'm not sure why or what they do.
>
> Anyway, at one of those sites, it takes a count of 15 or more for an
> outgoing call to connect. I know some delay is expected with that
> setup, but that's quite a bit longer than at our comparable sites.
>
> Is that length of delay still within expectations? Or is there
> something perhaps I can do to speed that up?
>
> Thanks
> jeff
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
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>
> ------------------------------
>
> Message: 32
> Date: Thu, 25 Jun 2009 08:57:39 -0500
> From: "Voice Noob" <<a href="mailto:voicenoob@gmail.com">voicenoob@gmail.com</a>>
> To: <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>,
> <<a href="mailto:ask-icd-ivr-support@external.cisco.com">ask-icd-ivr-support@external.cisco.com</a>>
> Subject: [cisco-voip] VoicemailQueuing
> Message-ID: <005401c9f59c$ed698a70$c83c9f50$@com>
> Content-Type: text/plain; charset="us-ascii"
>
> I am using the voicemail.aef and voicemailqueing.aef from this website.
>
> <a href="http://www.uccx.net/media/g/scriptexamples-5x/default.aspx?PageIndex=2">http://www.uccx.net/media/g/scriptexamples-5x/default.aspx?PageIndex=2</a>
>
>
>
> I have everything working well but have a few questions and hope someone can
> help me out. On the queuing aspect of the call when it gets presented to the
> agent they will press 2 and dial the original caller number. How can I setup
> up some type of logic so that if the remote party does not answer or the
> agent gets a voicemail box of the caller that the agent can hang-up and have
> the call wait a period of time and then get sent back to the queue to the
> agents. I guess I am looking for some type of interaction with the CAD
> software or even a DTMF entree to tell UCCX that the call was not handled
> and needs to be called again at a different time.
>
>
>
>
>
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>
> ------------------------------
>
> Message: 33
> Date: Thu, 25 Jun 2009 15:13:39 +0100
> From: Ian MacKinnon <<a href="mailto:Ian.Mackinnon@lumison.net">Ian.Mackinnon@lumison.net</a>>
> To: Jeff Ruttman <<a href="mailto:ruttmanj@carewisc.org">ruttmanj@carewisc.org</a>>, cisco-voip
> <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: Re: [cisco-voip] Slow to connect calls
> Message-ID:
>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Jeff,
> That sounds like a dial plan problem ie it is waiting for another digit, and then timing out.
>
> Can you dial the number before hitting dial on the phone so it is all present as opposed to lifting the handset and dialling each digit in turn?
>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jeff Ruttman
> Sent: 25 June 2009 14:47
> To: cisco-voip
> Subject: [cisco-voip] Slow to connect calls
>
> Greetings,
>
> Some of our sites have DID trunk ports and POTS lines, and we have MGCP controlled GWs with FXS and FXO configured. We also have for these sites H.323 GWs--which frankly I'm not sure why or what they do.
>
> Anyway, at one of those sites, it takes a count of 15 or more for an outgoing call to connect. I know some delay is expected with that setup, but that's quite a bit longer than at our comparable sites.
>
> Is that length of delay still within expectations? Or is there something perhaps I can do to speed that up?
>
> Thanks
> jeff
>
> CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
>
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>
> This email and any files transmitted with it are confidential and intended
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> ------------------------------
>
> Message: 34
> Date: Thu, 25 Jun 2009 11:17:01 -0400 (EDT)
> From: <a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>
> To: cisco-voip voyp list <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>>
> Subject: [cisco-voip] TAC confirms incorrect filename on CCO
> Message-ID: <<a href="mailto:B60138B2-E84E-4814-9A28-4C7F94F0089E@uoguelph.ca">B60138B2-E84E-4814-9A28-4C7F94F0089E@uoguelph.ca</a>>
> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
>
> For what it's worth, the TAC has confirmed the 7.1(2) CUC filename is
> incorrect on CCO.
>
> I mentioned this in an earlier post.
>
> Lelio Fulgenzi, Senior Analyst
> Computing & Communications
> University of Guelph
> 519-824-4120 x56354
>
> ...sent from my iPod - please pardon my fat fingers ;)
>
> [XKJ2000]
>
>
> ------------------------------
>
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>
>
> End of cisco-voip Digest, Vol 68, Issue 23
> ******************************************
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