[cisco-voip] upgrading CUCM from 6.1.2 to 6.1.3a

[Cristobal Priego]

Wes is right, on Version 6.X DNS and NTP are very sensitive, during an
upgrade/install you want to have those configured.
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

2009/6/25 Wes Sisk <wsisk at cisco.com>

> 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 <
> robert.knapp at spanlink.com> 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:
> cisco-voip-bounces at puck.nether.net [cisco-voip-bounces at puck.nether.net] On
> Behalf Of cisco-voip-request at puck.nether.net [
> cisco-voip-request at puck.nether.net] > Sent: Thursday, June 25, 2009 12:00
> PM > To: cisco-voip at puck.nether.net > Subject: cisco-voip Digest, Vol 68,
> Issue 23 > > Send cisco-voip mailing list submissions to >
> cisco-voip at puck.nether.net > > 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 >
> cisco-voip-request at puck.nether.net > > You can reach the person managing
> the list at > cisco-voip-owner at puck.nether.net > > 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 (
> lelio at uoguelph.ca) > > >
> ---------------------------------------------------------------------- > >
> Message: 1 > Date: Wed, 24 Jun 2009 13:20:31 -0400 > From: "Miller, Steve"
> <MillerS at DicksteinShapiro.COM> > To: cisco-voip at puck.nether.net > Subject:
> [cisco-voip] Has Anyone Run A Primary Unity Server on a > Physical Server
> and the Failover on a Virtual Server? > Message-ID: <
> 418329B7ED67E64BBD2BAA97078D70D001C62105 at DCEX2.DSMO.COM> > 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 > MillerS at dicksteinshapiro.com > > > >
> -------------------------------------------------------- > This e-mail
> message and any attached files are confidential and are intended solely for
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> If you are not the intended recipient or person responsible for delivering
> this confidential > communication to the intended recipient, you have
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> > > ------------------------------ > > Message: 2 > Date: Wed, 24 Jun 2009
> 12:43:35 -0500 > From: Tim Frazee <tfrazee at gmail.com> > To:
> cisco-voip at puck.nether.net > Subject: [cisco-voip] create a software CFB
> that got deleted > Message-ID: > <
> 30ce418a0906241043w2acc572fi4c612471267405b7 at mail.gmail.com> >
> 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? > -------------- next part --------------
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> > > ------------------------------ > > Message: 3 > Date: Wed, 24 Jun 2009
> 13:06:07 -0500 > From: Erick Bergquist <erickbee at gmail.com> > To:
> cisco-voip mailinglist <cisco-voip at puck.nether.net> > Subject:
> [cisco-voip] Call Manager 5.1.1a install file? > Message-ID: > <
> f4445faf0906241106q6f3c2ddbjb7f4f78be34357ce at mail.gmail.com> >
> 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 <asobihoudai at yahoo.com> > To: "Miller,
> Steve" <MillerS at DicksteinShapiro.COM>, > cisco-voip at puck.nether.net >
> Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on a >
> Physical Server and the Failover on a Virtual Server? > Message-ID: <
> 851685.56437.qm at web111314.mail.gq1.yahoo.com> > 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 at DicksteinShapiro.COM> > To:
> cisco-voip at puck.nether.net > 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 > MillerS at dicksteinshapiro.com > >
> -------------------------------------------------------- > 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
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> > > ------------------------------ > > Message: 5 > Date: Wed, 24 Jun 2009
> 14:28:32 -0400 > From: Jason Burns <burns.jason at gmail.com> > To: Erick
> Bergquist <erickbee at gmail.com> > Cc: cisco-voip mailinglist <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] Call Manager
> 5.1.1a install file? > Message-ID: > <
> 78d9bfc20906241128u76aa5207pc5bec4e836ce6b0e at mail.gmail.com> >
> 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 <
> erickbee at gmail.com> 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 >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > --------------
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> > > ------------------------------ > > Message: 6 > Date: Wed, 24 Jun 2009
> 14:39:05 -0400 > From: "Jason Aarons (US)" <jason.aarons at us.didata.com> >
> To: "Paul" <asobihoudai at yahoo.com>, "Miller, Steve" >
> <MillerS at DicksteinShapiro.COM>, <cisco-voip at puck.nether.net> > Subject:
> Re: [cisco-voip] Has Anyone Run A Primary Unity Server on > aPhysical Server
> and the Failover on a Virtual Server? > Message-ID: >
> <C1FE15183DA37645BC0633BC604E44F00F48FA41 at USNAEXCH.na.didata.local> >
> Content-Type: text/plain; charset="us-ascii" > > Design Guide for Cisco
> Unity Virtualization > >
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity/virtualization_desig >
> n/guide/cuvirtualdgx.html > > > > > > Check the archives for this list, it
> was discussed last month -jason > > > > From:
> cisco-voip-bounces at puck.nether.net > [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Paul > Sent: Wednesday,
> June 24, 2009 1:24 PM > To: Miller, Steve; cisco-voip at puck.nether.net >
> 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 at DicksteinShapiro.COM> > To: cisco-voip at puck.nether.net > 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 >
> MillerS at dicksteinshapiro.com > > > >
> -------------------------------------------------------- > > 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 > postmaster at dicksteinshapiro.com > > > > Dickstein Shapiro LLP >
> > http://www.DicksteinShapiro.com > > > >
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> > > ------------------------------ > > Message: 7 > Date: Wed, 24 Jun 2009
> 14:41:17 -0400 > From: Nick Matthews <matthnick at gmail.com> > To: Paul <
> asobihoudai at yahoo.com> > Cc: VOIP Group <cisco-voip at puck.nether.net> >
> Subject: Re: [cisco-voip] FW: CME web access disable > Message-ID: > <
> 56c3b48b0906241141s497c4e82l67cd5dbe7203c14c at mail.gmail.com> >
> 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: >
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090> > > -nick > > On Wed, Jun 24, 2009 at 10:15 AM, Paul<
> asobihoudai at yahoo.com> wrote: > > > Block port 80 on every switchport that
> has an IP phone on it. > > > > > ________________________________ > From:
> Ahmed Elnagar <ahmed_elnagar at hotmail.com> > To: VOIP Group <
> cisco-voip at puck.nether.net> > 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 > cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > > >
> ------------------------------ > > Message: 8 > Date: Wed, 24 Jun 2009
> 21:43:09 +0300 > From: Ahmed Elnagar <ahmed_elnagar at hotmail.com> > To: <
> matthnick at gmail.com>, <asobihoudai at yahoo.com> > Cc: VOIP Group <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] FW: CME web access
> disable > Message-ID: <BLU106-W13192812FAE56DC190EA5C87370 at 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: matthnick at gmail.com > To: asobihoudai at yahoo.com >
> CC: ahmed_elnagar at hotmail.com; cisco-voip at puck.nether.net > >
> telephony-service > service phone webAccess 1 > > > Then reset the phone. >
> > > Here's the full list of variables and their settings: >
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090> > > -nick > > On Wed, Jun 24, 2009 at 10:15 AM, Paul<
> asobihoudai at yahoo.com> wrote: > > > Block port 80 on every switchport that
> has an IP phone on it. > > > > > ________________________________ > From:
> Ahmed Elnagar <ahmed_elnagar at hotmail.com> > To: VOIP Group <
> cisco-voip at puck.nether.net> > 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 > cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > >
> _________________________________________________________________ > Show
> them the way! Add maps and directions to your party invites. >
> http://www.microsoft.com/windows/windowslive/products/events.aspx >
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> > > ------------------------------ > > Message: 9 > Date: Wed, 24 Jun 2009
> 14:16:42 -0500 > From: "Jeff Ruttman" <ruttmanj at carewisc.org> > To:
> "cisco-voip" <cisco-voip at puck.nether.net> > Subject: [cisco-voip] PRI
> Protocol NAT1, NAT2, custom? > Message-ID: > <
> 07365C3161D8D8419EE51C3834C02205B84D47 at ma1-exc01.ec2802.elderc.org> >
> 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)" <Matt.Slaga at us.didata.com> > To:
> Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip > <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] PRI Protocol NAT1,
> NAT2, custom? > Message-ID: >
> <5FE225375F6E3F4493474B90BC1B94EFB95160515B at 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:
> cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] 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
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> your computer. Thank you. > -------------- next part -------------- > An
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> > > ------------------------------ > > Message: 11 > Date: Wed, 24 Jun 2009
> 14:30:55 -0500 > From: "Jeff Ruttman" <ruttmanj at carewisc.org> > To: "Matt
> Slaga (US)" <Matt.Slaga at us.didata.com>, "cisco-voip" > <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] PRI Protocol NAT1,
> NAT2, custom? > Message-ID: > <
> 07365C3161D8D8419EE51C3834C02205B84D49 at ma1-exc01.ec2802.elderc.org> >
> 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:
> Matt.Slaga at us.didata.com] > 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: cisco-voip-bounces at puck.nether.net > [mailto:
> cisco-voip-bounces at puck.nether.net] 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,
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> > > ------------------------------ > > Message: 12 > Date: Wed, 24 Jun 2009
> 15:34:57 -0400 > From: "Matt Slaga (US)" <Matt.Slaga at us.didata.com> > To:
> Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip > <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] PRI Protocol NAT1,
> NAT2, custom? > Message-ID: >
> <5FE225375F6E3F4493474B90BC1B94EFB951605167 at 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:
> ruttmanj at carewisc.org] > 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:Matt.Slaga at us.didata.com] > 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:
> cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] 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
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> 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. > -------------- next part -------------- > An
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> > > ------------------------------ > > Message: 13 > Date: Wed, 24 Jun 2009
> 15:57:08 -0400 > From: Peter Slow <peter.slow at gmail.com> > To: Tim Frazee
> <tfrazee at gmail.com> > Cc: cisco-voip at puck.nether.net > Subject: Re:
> [cisco-voip] create a software CFB that got deleted > Message-ID: > <
> 53fc16d40906241257r19946bc3qcd6282ad951f7dbd at mail.gmail.com> >
> 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<tfrazee at gmail.com> 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 >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > > > >
> ------------------------------ > > Message: 14 > Date: Wed, 24 Jun 2009
> 23:12:35 +0300 > From: Dew Swen <dew.swen at gmail.com> > To: Mehmet Turunc <
> turunc.mehmet at gmail.com> > Cc: cisco-voip at puck.nether.net > Subject: Re:
> [cisco-voip] destination-pattern "T" question > Message-ID: > <
> ae5778960906241312j5eb22bf5g91fcc7abceea99d5 at mail.gmail.com> >
> 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 <turunc.mehmet at gmail.com>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 > cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > >
> -------------- next part -------------- > An HTML attachment was scrubbed...
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> > > ------------------------------ > > Message: 15 > Date: Wed, 24 Jun 2009
> 16:44:39 -0400 > From: "Jason Aarons (US)" <jason.aarons at us.didata.com> >
> To: <cisco-voip at puck.nether.net> > Subject: [cisco-voip] Does Unity
> Connection 7.1.2a support SIP RFC > 2833 for DTMF > Message-ID: >
> <C1FE15183DA37645BC0633BC604E44F00F48FCF1 at 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. > -------------- next part
> -------------- > An HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 16 > Date: Wed, 24 Jun 2009
> 12:49:31 -0700 > From: Tony Underwood <tony at cambiumdata.com> > To: Daniel
> <dan.voip at danofive.id.au>, "cisco-voip at puck.nether.net" > <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] Multicast MoH
> Delay > Message-ID: > <
> 0F205F18DCB4724DB15EAF8FF93E0A21129EAEE6FE at P3PW5EX1MB04.EX1.SECURESERVER.NET>
> > > 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 > http://www.cambiumdata.com<http://www.cambiumdata.com/> > >
> From: cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel > Sent: Wednesday,
> June 24, 2009 1:46 AM > To: cisco-voip at puck.nether.net > 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 --------------
> > An HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 17 > Date: Thu, 25 Jun 2009
> 08:44:12 +1000 > From: Daniel <dan.voip at danofive.id.au> > To: "
> cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net> > Subject: Re:
> [cisco-voip] Multicast MoH Delay > Message-ID: > <
> f861d63a0906241544t965b6c6x53696a9ddd691e7f at mail.gmail.com> >
> 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 <tony at cambiumdata.com>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 > > http://www.cambiumdata.com > > > >
> *From:* cisco-voip-bounces at puck.nether.net [mailto: >
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Daniel > *Sent:*
> Wednesday, June 24, 2009 1:46 AM > *To:* cisco-voip at puck.nether.net >
> *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: 18 > Date: Wed, 24 Jun 2009
> 15:00:37 -0700 (PDT) > From: Jake Doe <jd80301 at yahoo.com> > To:
> cisco-voip at puck.nether.net > Subject: [cisco-voip] SIP Route Pattern >
> Message-ID: <873114.16995.qm at web50806.mail.re2.yahoo.com> > 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
> -------------- > An HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 19 > Date: Wed, 24 Jun 2009
> 19:19:13 -0400 > From: "Jason Aarons (US)" <jason.aarons at us.didata.com> >
> To: <cisco-voip at puck.nether.net> > Subject: [cisco-voip] TAPS and UCCX
> 5.0.2 ? > Message-ID: >
> <C1FE15183DA37645BC0633BC604E44F00F4D14F7 at 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. > -------------- next part -------------- > An
> HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 20 > Date: Wed, 24 Jun 2009
> 20:34:25 -0400 > From: "Adam Frankel" <afrankel at cisco.com> > To: "'Jason
> Aarons \(US\)'" <jason.aarons at us.didata.com>, > <
> cisco-voip at puck.nether.net> > 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:
> cisco-voip-bounces at puck.nether.net > [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US) > Sent:
> Wednesday, June 24, 2009 4:45 PM > To: cisco-voip at puck.nether.net >
> 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. > > -------------- next part -------------- > An
> HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 21 > Date: Wed, 24 Jun 2009
> 19:45:42 -0500 > From: Erick Bergquist <erickbee at gmail.com> > To: Jason
> Burns <burns.jason at gmail.com> > Cc: cisco-voip mailinglist <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] Call Manager
> 5.1.1a install file? > Message-ID: > <
> f4445faf0906241745q7b616ea6p3c44f01d8374fd6 at mail.gmail.com> >
> Content-Type: text/plain; charset=ISO-8859-1 > > Yea, thanks. > > On Wed,
> Jun 24, 2009 at 1:28 PM, Jason Burns<burns.jason at gmail.com> wrote: > > >
> That would probably be 5.1.1.2000-1 or 2. > > On Wed, Jun 24, 2009 at 2:06
> PM, Erick Bergquist <erickbee at gmail.com> 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 > cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > > > >
> ------------------------------ > > Message: 22 > Date: Thu, 25 Jun 2009
> 11:19:57 +1000 > From: ciscozest <ciscozest at gmail.com> > To: cisco-voip
> mailinglist <cisco-voip at puck.nether.net> > Subject: [cisco-voip] T.37 Fax
> Redundancy > Message-ID: > <
> f99cc3f60906241819l32750916n541060c9184d049e at mail.gmail.com> >
> 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 <tony at cambiumdata.com> > To: Daniel
> <dan.voip at danofive.id.au>, "cisco-voip at puck.nether.net" > <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] Multicast MoH
> Delay > Message-ID: > <
> 0F205F18DCB4724DB15EAF8FF93E0A21129EAEE76A at P3PW5EX1MB04.EX1.SECURESERVER.NET>
> > > 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 > http://www.cambiumdata.com<
> http://www.cambiumdata.com/> > > From: cisco-voip-bounces at puck.nether.net[mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel > Sent: Wednesday,
> June 24, 2009 5:44 PM > To: cisco-voip at puck.nether.net > 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 <tony at cambiumdata.com<mailto:tony at cambiumdata.com>>
> 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 > >
> http://www.cambiumdata.com<http://www.cambiumdata.com/> > > > > From:
> cisco-voip-bounces at puck.nether.net<mailto:
> cisco-voip-bounces at puck.nether.net> [mailto:
> cisco-voip-bounces at puck.nether.net<mailto:
> cisco-voip-bounces at puck.nether.net>] On Behalf Of Daniel > Sent:
> Wednesday, June 24, 2009 1:46 AM > To: cisco-voip at puck.nether.net<mailto:
> cisco-voip at puck.nether.net> > 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 at DicksteinShapiro.COM> >
> To: cisco-voip at puck.nether.net > Subject: [cisco-voip] One User Cannot Be
> Dialed By Name > Message-ID: <
> 418329B7ED67E64BBD2BAA97078D70D001C62118 at DCEX2.DSMO.COM> > 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 > MillerS at dicksteinshapiro.com > > > >
> -------------------------------------------------------- > 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 postmaster at dicksteinshapiro.com >
> > Dickstein Shapiro LLP > http://www.DicksteinShapiro.com > >
> ==============================================================================
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> > > ------------------------------ > > Message: 25 > Date: Wed, 24 Jun 2009
> 20:00:39 -0700 > From: Cristobal Priego <cristobalpriego at gmail.com> > To:
> "Miller, Steve" <MillerS at DicksteinShapiro.COM> > Cc: "
> cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net> > Subject: Re:
> [cisco-voip] One User Cannot Be Dialed By Name > Message-ID: <
> 3E4765E5-CCD9-485C-8FC9-949F63DB0E93 at gmail.com> > 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 at 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 >
> MillerS at dicksteinshapiro.com > > >
> -------------------------------------------------------- > 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
> postmaster at dicksteinshapiro.com > > Dickstein Shapiro LLP >
> http://www.DicksteinShapiro.com > > === > === > === >
> ===================================================================== >
> _______________________________________________ > cisco-voip mailing list >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > --------------
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> > > ------------------------------ > > Message: 26 > Date: Wed, 24 Jun 2009
> 23:44:08 -0400 > From: Dustin S Fowler <dustin.s.fowler at gmail.com> > To:
> "Jason Aarons (US)" <jason.aarons at us.didata.com> > Cc: "
> cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net> > Subject: Re:
> [cisco-voip] TAPS and UCCX 5.0.2 ? > Message-ID: > <
> f2d16ce0906242044r1ce0dc65jd1eb8c3d31375e05 at mail.gmail.com> >
> 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) < >
> jason.aarons at us.didata.com> 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 > 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. * > >
> _______________________________________________ > cisco-voip mailing list >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > >
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> > > ------------------------------ > > Message: 27 > Date: Wed, 24 Jun 2009
> 21:07:31 -0700 > From: Mark Holloway <mh at markholloway.com> > To: Jason
> Aarons (US) <jason.aarons at us.didata.com> > Cc: cisco-voip at puck.nether.net> Subject: Re: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC >
> 2833 for DTMF > Message-ID: <
> F6250C96-D4DD-4FAD-BDCC-E64DF7B9C442 at markholloway.com> > 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 > 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. > >
> _______________________________________________ > cisco-voip mailing list >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > --------------
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> > > ------------------------------ > > Message: 28 > Date: Thu, 25 Jun 2009
> 02:59:33 -0300 > From: "ROJAS, Mario" <Mario.ROJAS at LA.LOGICALIS.COM> > To:
> <cisco-voip at puck.nether.net> > Subject: [cisco-voip] Show Saved Enterprise
> Data in CAD > Message-ID: > <
> 1081D526E13AB442851D3AE541A2F112CBBB8C at SNLAR-EXCH01.LA.LOGICALIS.COM> >
> 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:mario.rojas at la.logicalis.com>
> mario.rojas at la.logicalis.com > <http://www.la.logicalis.com>
> www.la.logicalis.com > <http://www.logicalisnow.com/> www.logicalisnow.com> > > 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. > La modificaci?n,
> retransmisi?n, difusi?n, copia u otro uso de esta informaci?n por cualquier
> medio, por personas distintas al destinatario, est?n estrictamente
> prohibidas. > > 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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> > > ------------------------------ > > Message: 29 > Date: Thu, 25 Jun 2009
> 11:17:58 +0300 > From: Mehmet Turunc <turunc.mehmet at gmail.com> > To: Dew
> Swen <dew.swen at gmail.com> > Cc: cisco-voip at puck.nether.net > Subject: Re:
> [cisco-voip] destination-pattern "T" question > Message-ID: > <
> 7d505d120906250117n274614c7j16d6148da9a1102b at mail.gmail.com> >
> 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 <
> dew.swen at gmail.com> 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 <turunc.mehmet at gmail.com>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 > cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > > >
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> > > ------------------------------ > > Message: 30 > Date: Thu, 25 Jun 2009
> 08:37:29 -0500 > From: "Beck, Christopher" <CBeck at usg.com> > To: "ROJAS,
> Mario" <Mario.ROJAS at LA.LOGICALIS.COM>, > "cisco-voip at puck.nether.net" <
> cisco-voip at puck.nether.net> > Subject: Re: [cisco-voip] Show Saved
> Enterprise Data in CAD > Message-ID: > <
> AF8E5A5BA2DF074E8B46A3B4A8BC265513806156 at CERO-MB-01.USG.NET> >
> 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) >
> cbeck at usg.com > > From: cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of ROJAS, Mario > Sent:
> Thursday, June 25, 2009 1:00 AM > To: cisco-voip at puck.nether.net >
> 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 > mario.rojas at la.logicalis.com<mailto:
> mario.rojas at la.logicalis.com> > www.la.logicalis.com<
> http://www.la.logicalis.com> > www.logicalisnow.com<
> http://www.logicalisnow.com/> > > Por favor, piense en el medioambiente
> antes de imprimir este email. > La presente informaci?n se env?a ?nicamente
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> ESET Smart Security, version of virus signature database 4187 (20090625)
> __________ > > The message was checked by ESET Smart Security. > >
> http://www.eset.com > > > 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
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> 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. > -------------- next part -------------- > An
> HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 31 > Date: Thu, 25 Jun 2009
> 08:47:04 -0500 > From: "Jeff Ruttman" <ruttmanj at carewisc.org> > To:
> "cisco-voip" <cisco-voip at puck.nether.net> > Subject: [cisco-voip] Slow to
> connect calls > Message-ID: > <
> 07365C3161D8D8419EE51C3834C02205B84D50 at ma1-exc01.ec2802.elderc.org> >
> 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. > -------------- next part -------------- > An
> HTML attachment was scrubbed... > URL: <
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> > > ------------------------------ > > Message: 32 > Date: Thu, 25 Jun 2009
> 08:57:39 -0500 > From: "Voice Noob" <voicenoob at gmail.com> > To: <
> cisco-voip at puck.nether.net>, > <ask-icd-ivr-support at external.cisco.com> >
> 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. > >
> http://www.uccx.net/media/g/scriptexamples-5x/default.aspx?PageIndex=2 > >
> > > 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 <Ian.Mackinnon at lumison.net> > To:
> Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip > <
> cisco-voip at puck.nether.net> > 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: cisco-voip-bounces at puck.nether.net[mailto:
> cisco-voip-bounces at puck.nether.net] 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. > >
> ________________________________ > -- > > This email and any files
> transmitted with it are confidential and intended > solely for the use of
> the individual or entity to whom they are addressed. > If you have received
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> excepted. Please note that any views or opinions > presented in this email
> are solely those of the author and do not > necessarily represent those of
> Lumison. > Finally, the recipient should check this email and any
> attachments for the > presence of viruses. Lumison accept no liability for
> any > damage caused by any virus transmitted by this email. > --------------
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> > > ------------------------------ > > Message: 34 > Date: Thu, 25 Jun 2009
> 11:17:01 -0400 (EDT) > From: lelio at uoguelph.ca > To: cisco-voip voyp list
> <cisco-voip at puck.nether.net> > Subject: [cisco-voip] TAC confirms
> incorrect filename on CCO > Message-ID: <
> B60138B2-E84E-4814-9A28-4C7F94F0089E at uoguelph.ca> > 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] > > >
> ------------------------------ > >
> _______________________________________________ > cisco-voip mailing list >
> cisco-voip at puck.nether.net >
> https://puck.nether.net/mailman/listinfo/cisco-voip > > > End of
> cisco-voip Digest, Vol 68, Issue 23 >
> ****************************************** >
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