[cisco-voip] IPCC Record Document variable
Ed Leatherman
ealeatherman at gmail.com
Wed Nov 19 16:02:47 EST 2008
Hi Chris,
I would recommend taking a look at the UCCX script repository, there are
some great examples in there and I think there is a script there that does
something similar that you could pull from.
www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/design/guide/script_repository.zip
Ed
On Wed, Nov 19, 2008 at 2:51 PM, Chris Hill <chris.hill at swri.org> wrote:
> I am working on setting up a way in which our helpdesk can change the
> welcome message to a temp emergency message.
>
> I thought of using the record variable and then choosing the welcome menu
> message to play the recording document variable created with the record
> variable. This would work but my question is:
> Will this variable get flushed out or purged if so at what time or event
> will this variable be emptied?
>
> Any advice or suggestions on this greatly appreciated.
>
>
>
> Chris Hill
> Information Technology Center
> Southwest Research Institute
> Phone: (210) 522-3405
> chill at swri.org
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> cisco-voip-request at puck.nether.net
> Sent: Wednesday, November 19, 2008 12:54 PM
> To: cisco-voip at puck.nether.net
> Subject: cisco-voip Digest, Vol 61, Issue 131
>
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> Today's Topics:
>
> 1. Re: Tac Guys...... CM 4.1 and headset hookswitch control
> (CM5.1(3)) (Scott Voll)
> 2. MGCP gateway 2 dialing (cisco.voip)
> 3. Re: MGCP gateway 2 dialing (Wes Sisk)
> 4. Re: Verifying load ... (Wes Sisk)
> 5. Re: MGCP gateway 2 dialing (cisco.voip)
> 6. CUCM 6.1.(2) upgrade and phone loads (Maniccia, Thomas)
> 7. Re: ilbc codec and DevicePool for a Transcoder? (Peter Slow)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 19 Nov 2008 09:51:39 -0800
> From: "Scott Voll" <svoll.voip at gmail.com>
> Subject: Re: [cisco-voip] Tac Guys...... CM 4.1 and headset hookswitch
> control (CM5.1(3))
> To: "Dana Tong (AU)" <Dana.Tong at didata.com.au>,
> "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID:
> <f84a38d30811190951i1652f859vf350fd14f981a45f at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> on cm 6.12 it's at the very bottom of the device page.
>
> I also believe you need a newer firmware. 8.3.x or maybe better and on CM
> 5.1 you might need a ES / SR to get the feature.
>
> Scott
>
> On Tue, Nov 18, 2008 at 10:31 PM, Dana Tong (AU)
> <Dana.Tong at didata.com.au>wrote:
>
> > I'm in Australia but the customer's system that I'm looking at doesn't
> > appear to have any other locale installed bar the default American. Would
> > this stop the phone having the option of the headset hookswitch? Why
> won't
> > the EHS work in America?
> >
> >
> > ------------------------------
> > *From:* Scott Voll [svoll.voip at gmail.com]
> > *Sent:* Wednesday, 19 November 2008 3:35 PM
> > *To:* Dana Tong (AU)
> > *Cc:* Jonathan Charles; cisco-voip at puck-nether.net
> >
> > *Subject:* Re: [cisco-voip] Tac Guys...... CM 4.1 and headset hookswitch
> > control (CM5.1(3))
> >
> > What model of headset? I can't remember model numbers but we started
> > w/ (i think) a 1110 and it does not work in the americas, only works in
> > europe. had to use the 9130 (again, I think it's the model number).
> >
> > you also have to turn on the HHC on the phone config page. Had it
> working
> > in 4.x and now in 6.x.
> >
> > Let us know.
> >
> > Scott
> >
> > On Tue, Nov 18, 2008 at 9:32 PM, Dana Tong (AU)
> <Dana.Tong at didata.com.au>wrote:
> >
> >> Have also tried phone firmware 8-4-1.
> >>
> >> Cheers
> >>
> >> ------------------------------
> >> *From:* cisco-voip-bounces at puck.nether.net [
> >> cisco-voip-bounces at puck.nether.net] On Behalf Of Dana Tong (AU) [
> >> Dana.Tong at didata.com.au]
> >> *Sent:* Wednesday, 19 November 2008 3:25 PM
> >> *To:* Jonathan Charles; Scott Voll
> >> *Cc:* cisco-voip at puck-nether.net
> >> *Subject:* Re: [cisco-voip] Tac Guys...... CM 4.1 and headset hookswitch
> >> control (CM5.1(3))
> >>
> >> Hey all,
> >>
> >> Has anyone been able to get the Jabra headsets with the HHC working on
> >> CallManager 5?
> >>
> >> This system is:
> >> CUCM version - 5.1.3.2000-3
> >> IP Phone - CP-7962G
> >> Phone firmware - SCCP42.8-3-3S
> >>
> >> URLs that have given me some information:
> >>
> >>
>
> http://www.jabra.com/sites/Jabra/GNImages/Campaigns/Cisco/FAQ_Cisco_low_2901
> .pdf<http://www.jabra.com/sites/Jabra/GNImages/Campaigns/Cisco/FAQ_Cisco_low_2901.pdf>
> >>
> >>
> >>
>
> http://www.jabra.com/Sites/Jabra/na-us/headsetsupport/pages/Knowledgebase.as
> px?kbid=916a5151-f671-46ad-8677-8d460ac111eb<http://www.jabra.com/Sites/Jabra/na-us/headsetsupport/pages/Knowledgebase.aspx?kbid=916a5151-f671-46ad-8677-8d460ac111eb>
> >>
> >> I believe that the phone is connected ok but just don't seem to be able
> to
> >> get it to work.
> >>
> >> Thanks
> >> Dana
> >>
> >>
> >>
> >> ------------------------------
> >> *From:* cisco-voip-bounces at puck.nether.net [
> >> cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles [
> >> jonvoip at gmail.com]
> >> *Sent:* Friday, 22 February 2008 2:54 AM
> >> *To:* Scott Voll
> >> *Cc:* cisco-voip at puck-nether.net
> >> *Subject:* Re: [cisco-voip] Tac Guys...... CM 4.1 and headset hookswitch
> >> control
> >>
> >> This message has been archived.
> >> So, what about CCM 5.1.3 and 7961s? Jonathan On Fri, Feb 15, 2008 at
> 3:29
> >> PM, Scott Voll <svoll.voip at gmail.com> wrote: > yes. sr6a added it...
> sr5b
> >> did not have it. > > Thanks > > scott > > > > On Fri, Feb 15, 2008 at
> 1:26
> >> PM, Wes Sisk <wsisk at cisco.com> wrote: > > > > > So you have the
> >> checkbox/dropdown now? > > > > /Wes > > > > > > > > > > Scott Voll
> wrote:
> >
> >> > > > to come full around on this. I just got off the phone with Jabra.
> >
> >
> >> > > I had all the requiremen
> >>
> >>
> >>
>
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> **
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> ------------------------------
>
> Message: 2
> Date: Wed, 19 Nov 2008 13:01:15 -0500
> From: "cisco.voip" <cisco.voip at verizon.net>
> Subject: [cisco-voip] MGCP gateway 2 dialing
> To: <cisco-voip at puck.nether.net>
> Message-ID: <000401c94a70$e74151a0$6c986bc0 at cs.myharris.net>
> Content-Type: text/plain; charset=iso-8859-1
>
> How do I implement two stage dialing on an MGCP gateway.
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 19 Nov 2008 13:12:50 -0500
> From: Wes Sisk <wsisk at cisco.com>
> Subject: Re: [cisco-voip] MGCP gateway 2 dialing
> To: "cisco.voip" <cisco.voip at verizon.net>
> Cc: cisco-voip at puck.nether.net
> Message-ID: <49245722.7000105 at cisco.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> if the mgcp gateway uses cucm as the call agent it is not possible. /wes
>
> On Wednesday, November 19, 2008 1:01:15 PM, cisco.voip
> <cisco.voip at verizon.net> wrote:
> > How do I implement two stage dialing on an MGCP gateway.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 19 Nov 2008 13:13:15 -0500
> From: Wes Sisk <wsisk at cisco.com>
> Subject: Re: [cisco-voip] Verifying load ...
> To: "cisco.voip" <cisco.voip at verizon.net>
> Cc: cisco-voip at puck.nether.net
> Message-ID: <4924573B.6040008 at cisco.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> phone is still performing TFTP download during that time.
>
> On Wednesday, November 19, 2008 12:39:19 PM, cisco.voip
> <cisco.voip at verizon.net> wrote:
> > Does anyone know why some phones are constantly 'Verifiying Load' before
> > activating?
> >
> > Does any know exactly what the phone is doing during this time. Is is
> > downloading the files again?
> >
> > Tx
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 19 Nov 2008 13:33:07 -0500
> From: "cisco.voip" <cisco.voip at verizon.net>
> Subject: Re: [cisco-voip] MGCP gateway 2 dialing
> To: "Wes Sisk" <wsisk at cisco.com>
> Cc: cisco-voip at puck.nether.net
> Message-ID: <000b01c94a75$4688a880$6c986bc0 at cs.myharris.net>
> Content-Type: text/plain; charset=iso-8859-1
>
> but with h323 gw it is possbile with the dial peer?
>
> ----- Original Message -----
> From: "Wes Sisk" <wsisk at cisco.com>
> To: "cisco.voip" <cisco.voip at verizon.net>
> Cc: <cisco-voip at puck.nether.net>
> Sent: Wednesday, November 19, 2008 1:12 PM
> Subject: Re: [cisco-voip] MGCP gateway 2 dialing
>
>
> > if the mgcp gateway uses cucm as the call agent it is not possible. /wes
> >
> > On Wednesday, November 19, 2008 1:01:15 PM, cisco.voip
> > <cisco.voip at verizon.net> wrote:
> > > How do I implement two stage dialing on an MGCP gateway.
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 19 Nov 2008 13:36:28 -0500
> From: "Maniccia, Thomas" <maniccia at buffalo.edu>
> Subject: [cisco-voip] CUCM 6.1.(2) upgrade and phone loads
> To: "'cisco-voip at puck.nether.net'" <cisco-voip at puck.nether.net>
> Message-ID:
>
> <09665E61F9A441438477E276F2810C0EBCED7370B1 at MBCCR2.itorg.ad.buffalo.edu>
>
> Content-Type: text/plain; charset="us-ascii"
>
>
> I will be upgrading from CUCM 6.1(1a) to 6.1.(2). Should the new default
> phone loads be placed on the nodes in the current 6.1(1) active partition
> or
> uploaded after the upgrade to 6.1(2) is complete and that is the active
> partition?
>
> TIA,
> Tom
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 19 Nov 2008 13:53:32 -0500
> From: "Peter Slow" <peter.slow at gmail.com>
> Subject: Re: [cisco-voip] ilbc codec and DevicePool for a Transcoder?
> To: akoul at dataways.gr
> Cc: cisco-voip at puck.nether.net
> Message-ID:
> <53fc16d40811191053o1213e946i652f334466fb8397 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Anthony, we need some more infromation to tell you waht's happening.
>
> can both devices do iLBC? Trying to undesrtand what call flow is
> requiing a transcoder.
> Where is your transcoder physically located?
>
> what types of devices are involved in the call?
> what codecs do the devices at the central site use and what are the
> model numbers of the phones in use? they don't all support iLBC.
> what device pool/region is the transcoder in and what is configured
> between that region and the central and remote site regions?
>
> -Peter
>
> On Wed, Nov 19, 2008 at 12:25 PM, Anthony Kouloglou <akoulgr at gmail.com>
> wrote:
> > Hi,
> > 1. i have configured an Enhanced IOS transcoder to be used by several
> remote
> > devices that want to use iLBC with the central site.
> > 2. The sccp transcoder is registered with the CUCM 6.1 ok. (ilbc is
> included
> > in the sccp profile)
> > 3. I have configured 2 regions one for the central site and one for the
> > remote. Audio codec between them is iLBC.
> > 4. I have configured a media resource group containing only the above
> > transcoder.
> > 5. I have also configured 2 device pools: each pool has one region of the
> > above (step 3)
> >
> > What must the media resource list be like in order for the configuration
> to
> > work?
> > What is the meaning of assigning a device pool inside the transcoder
> since
> > the transcoder is in a MRGL which inturn is used by a device pool?
> >
> > Thanks a lot
> > Anthony
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> ------------------------------
>
> _______________________________________________
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> cisco-voip at puck.nether.net
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>
>
> End of cisco-voip Digest, Vol 61, Issue 131
> *******************************************
>
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--
Ed Leatherman
Assistant Director, Voice Services
West Virginia University
Telecommunications and Network Operations
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