This is my recommendation.
Step 1) Create your Keynote presentation.
Step 2) Record your speaking part (separately) with an audio recording program.
I recommend Audacity. It's a great free application. You can download Audacity from www.sorceforge.com among other places. You'll also want to get the Audacity LIB file that is needed to export your audio recording to .mp3.
3) From the Keynote Inspector, under the Document tab, go to the audio section and add your new voice recording file to the Soundtrack.
4) From Keynote, go to File, Record Slideshow.
While the audio plays, click through your slides. This is getting the slide change time down.
5) From Keynote, go to File, Export. Make sure to have the following settings:
Playback Uses: Recorded Timing
Include audio (sound files, movie audio): NOT SELECTED
Include the slideshow soundtrack: NOT SELECTED
Include the slideshow recording: SELECTED!!!
Click next and your Keynote presentation should export to a .mov file (keep in mind, this .mov file has no audio. we'll take care of that in the last steps).
6)Open the new .mov Keynote presentation in Quicktime Pro.
7) Open the original .mp3 sound file in Quicktime Pro.
8) Select All of the .mp3 file and then click CTRL + C to copy it.
9) Go back to the .mov Keynote file and Select All. Then go up to Edit > Add to Movie.
This adds your sound and movie file together.
10) Finally click on File > Save As. Select, Save As A Self-Contained Movie and you're done.
I Hope This Helps!

Similar Messages

Fax outdial retries consume all voice channels on SIP 484 error (Cisco 2911)

I've been seeing a nasty fax/VoIP problem on a 2911, running IOS 15.0(1r)M12. Any suggestions would be welcome.
I have a 2911 which is set up to do T.37 offramp fax delivery (SMTP message is sent to 2911, which places a VoIP call over SIP/RTP/T.38 to deliver the fax). The mainline case is set up, and working correctly - faxes are delivered without issue. If a destination address is selected such that the VoIP switch returns a SIP 484 error, then everything fails in a spectacular fashion:
The outdial is immediately retried, placing another SIP INVITE to the switch, with the same destination address, which obviously also gets the same 484 response.
Each time the outdial takes place, it consumes voice channels on the DSP, which are not released on receipt of the 484.
When there are no free voice channels, a no circuit (0x22) error is returned, and all the voice channels are finally released.
The MTA that submitted the SMTP message retries every minute (it doesn't get a permanent failure report when the 2911 fails to place the call)
This leads to a situation where no fax calls can be placed, as all the voice channels are being used up by retrying this call that can never succeed.
Some other relevant information:
The VoIP switch does not return a 484 immediately. First it sends a SIP 183, and plays early media (an announcement about how the call isn't allowed).
It takes 8 seconds before the 484 is returned. The 2911 sends a new SIP INVITE every 8 seconds (as soon as it gets a 484 for the previous attempt).
The "sip-ua" statistics show that the INVITE retry counter is not being incremented (i.e. this is not a retry at the scope of the SIP stack).
The T1 cable is looped-back to the 2911, so that the complete path for fax delivery looks like this:
MTA ---SMTP---> 2911 ---T1---> 2911 ---SIP---> VoIP switch
If I set "mta receive generate permanent-error", then I still see this retry behaviour, with all the voice channels being consumed. Once that has happened (after about 3 minutes) the MTA does get the error response, and no longer retries every minute after that (although this setting has other negative effects that I'd like to avoid).
Does anyone have any idea how I can get the 2911 to return a permanent failure to the MTA after just a single outdial has failed with a SIP 484?
Here is the dial-peer config:
dial-peer voice 1 voip
translation-profile incoming IncomingVoip
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711ulaw
no vad
dial-peer voice 2 pots
destination-pattern ^0005
port 1/1:23
forward-digits all
dial-peer voice 3 pots
translation-profile incoming IncomingPRI_1_0
service onramp-app
incoming called-number ^0005
direct-inward-dial
port 1/0:23
dial-peer voice 4 mmoip
service fax_on_vfc_onramp_app out-bound
destination-pattern .
information-type fax
session target mailto:$m$@<DOMAIN NAME>
image encoding MH
dial-peer voice 101 mmoip
translation-profile incoming IncomingMMoIP
service offramp-app
information-type fax
incoming called-number .
dial-peer voice 102 pots
destination-pattern .
port 1/0:23
forward-digits all
dial-peer voice 103 pots
translation-profile incoming IncomingPRI_1_1
incoming called-number ^0007
direct-inward-dial
port 1/1:23
dial-peer voice 104 voip
translation-profile outgoing OutgoingVoip
destination-pattern ^0008
session protocol sipv2
session target ipv4:<VoIP SWITCH IP ADDRESS>
voice-class codec 1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 3 hs-redundancy 0 fallback pass-through g711ulaw
no vad

Hi Ellad.
Why don't try to use the 2811 as a SIP signalling proxy only?
In this way the media (RTP or T.38) will be handled only from the two MERA SoftSwitch.
To do this you must enable CUBE on your 2811 and use these special commands:
voice service voip
     media flow-around
     allow-connections sip to sip
     signaling forward unconditional
     sip
       rel1xx disable
       header-passing
       midcall-signaling passthru
       pass-thru headers unsupp
       pass-thru content unsupp
       pass-thru content sdp
I don't remember if we have already try this solution.
Regards.

Docs on adding new channels/portlats

I'm working with a group of people who are evaluating the 2005Q4 portal server. We want to add some new channels and maybe a customized portlet. I'm trying to narrow down the documents we need to read to figure out how to do this.
Can someone direct me to the correct documentation for this? It isn't possible for us to read every page of the portal server documentation so I need to narrow it down to what is necessary to add new channels.
So far I think the information may be in one of these two docs:
http://docs.sun.com/app/docs/doc/819-4156
http://docs.sun.com/app/docs/doc/819-4157

Here is a link that describes adding a portlet to a page on the desktop.
http://docs.sun.com/source/819-4154/dktopadm.html#wp26726
The anonymous portal desktop is defined under the authlessanonymous user.
The most common container that is used is the JSPTabContainer. Adding a tab is just like adding a channel, except that you choose the JSPTableContainerProvider for the Tab and add it to the JSPTabContainer.
The first tab on the anonymous user's desktop is called the MyFrontPageTabPanelContainer.
HTH,
Jim

Nokia CS-15: Deaktivating voice channel

Hello,
I searched the whole internet without find a solution, so I am now here...
Last week I bought the Nokia CS-15 Internet Stick in order to use it for mobile connections with my Vodafone UltraCard (it's a MultiSim). Unfortunately the CS-15 blocks the data AND voice channel while being connected. Because of this I can't receive telephone calls on my mobile while using the CS-15.
For other devices I found several AT commands to deactivate the voice channel, but no resolution for the CS-15. The known AT commands did not work on the Nokia device.
Have someone of you an idea to fix this problem?
Why does the stick block the voice channel?
Thanks a lot
Thomas

The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect VoIP calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful SIP proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
http://www.cisco.com/en/US/products/ps6441/prod_troubleshooting_guide09186a00801fa207.html#wp2077496

Adding a channel?

We are looking to subscribe to Mediaset Italia channel, but it is not currently offered. All other competitors have it. Will Verizon be adding this channel soon?

We do take customer input into account when making additions to the service. This would be a perfect candidate for the Verizon Idea Exchange. Please search on the idea exchange and cast your vote. There may be another member who made the same suggestion. If not, please add a new idea so your fellow community members can comment and vote on your suggestion. http://forums.verizon.com/t5/Verizon-Idea-Exchange/ct-p/ideaexchange . The Verizon Idea Exchange is there for you to submit your ideas and suggestions to Verizon so that we can consider them as we innovate our products and services.
Anthony_VZ
**If someones post has helped you, please acknowledge their assistance by clicking the red thumbs up button to give them Kudos. If you are the original poster and any response gave you your answer, please mark the post that had the answer as the solution**
Notice: Content posted by Verizon employees is meant to be informational and does not supersede or change the Verizon Forums User Guidelines or Terms or Service, or your Customer Agreement Terms and Conditions or plan

ANy MIB to poll for no. of active voice channel on 2611 with T1 Module

Hi ppl,
I have a 2611 router with a T1 voice card (NM-HDV-1T1-24) as a voice gateway in my office. IOS version c2600-is-mz.122-11.T2.bin has been loaded and running.
I have tried to poll the mib oject "cvActiveCallConnectionID " under the Cisco-Voice-Dial-Control MIB for no. of active voice channel. I use solarwinds engineering edition 2002 to poll this device. It shows an error said that the mib is not supported in this version of IOS. We are running VoFR in our existing environment
I have checked in the Cisco MIB webpage --
ftp://ftp.cisco.com/pub/mibs/supportlists/c2611/c2611-supportlist.html
-- that the IOS is supported in version 11.3!!! Is it only support in 11.3 or any later version?
If not, any possible mib or workaround can solve my problem.
Regards,
Sam

The information is in the mib object cvCallActiveTable. There is no frame-relay equivalent to the VoIP object. The only frame-relay specific SNMP info you can verify is CISCO-VOICE-FR-DIAL-CONTROL-MIB

Video jerky after adding voice-over

The video imports nicely, but when I do voice-over using a Logitech headphone, the video portion becomes jerky. Something similar was discussed in the forum about a year ago, but it appears that nothing was resolved? Would appreciate help on this. Thanks
Betty
iMacG5 Mac OS X (10.3.9) iMovie 5.0.2 QT 7.1.3

always convert and AIFF files to MP3 first..... only have about 20 audio clips, but they're mostly voice over and AIFF files.
No. iMovie converts MP3 format to AIFF anyway. Some have found more stuttering problems with voice-overs.
I'm stuck right now with a half hour video that i managed to finishe editing but can't play or export to tape without terrible stuttering
If you have enough space on your hard drive, you can 'share' your finished movie as QuickTime Full Quality. Import this QT movie into a new iMovie project. It should now play without stuttering, allowing you to export it back to DV tape. If the QT movie plays fine, you can delete the original iMovie.
You can add chapter markers and use this version for iDVD.
I use this method all the time. Sometimes I have had to create QT movie version three times when editing a long or complex iMovie. Even before adding audio, my movies will skip/stutter on playback if I have lots of transitions/effects/total clips. I have learned to edit the movie by cropping the video clips first, adding stills, doing transitions and titles/effects, and adding the audio last. If it starts to skip with the addition of the first audio clip, I delete that clip, share as quicktime, and then work on the quicktime version in a new iMovie and go ahead with adding the audio to that.

Easiest solution for adding Fibre Channel Tape Backup to Xserve RAID setup

I'm interested in getting a Fibre Channel LTO-3 tape library such as the Exabyte 221L for my Xserve RAID/Xserve G5 setup. I realize there are SCSI LTO-3 solutions that are less expensive than FC, but I'd prefer not to have to add a SCSI card to my Xserve, mainly because I'd have to pull the video card and run the server headless (horrors!). However, I understand that I must get a Fibre Channel switch to allow the LTO library, the Xserve, and the Xserve RAID to communicate... you can't just daisy chain FC devices. The least expensive switch I can find is the new QLogtic SANbox 3050 for about $1800. For my situation, in which these will be the only FC devices I'll probably ever use, does this switch sound appropriate? The specs on the switch can be found here: http://www.qlogic.com/products/sanbox/sanbox_3050.asp.
Xserve RAID & Xserve G5 Mac OS X (10.4.5)

For simple environments like yours, it's trivially easy. You must plug everything in, and you're good to go... as simple as a network switch.
Now if you start adding multiple hosts (i.e. plug in a second Xserve or G5), you need to be careful, because in the default configuration everything is "wide open," so every device can see every storage LUN. This would actually be problematic, as systems could (actually, would) corrupt each other's storage. When you get to this point, you'll need to look at either zoning the ports on the switch, or using LUN masking on the Xserve RAID to allocate specific pieces of storage to specific hosts.
But for now, just get the switch, plug everything in, and you're off.
BTW if you mark "answered" for me, versus for the thread overall, it helps me out a bit in this forum ranking system, whatever it is.
Cheers.

How to verify voice channels on voice gateway

Hello,
I am in the testing phase of deploying a 2921 voice gateway with 2 PRIs. After connecting the T1s, how do I verify if all the channels are up and ready for use? Also, how do I verify if clocking on the T1 interfaces are configured correctly?
This 2921 is a replacement for a 2811 currently being used. When I login to the 2811 via SSH, I see this:
560240: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560241: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560242: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560243: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560244: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560245: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560246: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560247: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560248: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560249: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560250: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
560251: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
How do I do that with the 2921 in an SSH session? I've enabled "logging monitor" and "term mon" but I still can't see messages like that.
I thank you very much for your time.

Hello,
To check clocking you can use the "show network-clock" command. For example:
SiteA-RTR#show network-clock
Network Clock Configuration
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
10 Backplane GOOD PLL
Current Primary Clock Source
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
The above is an example of a "good" configuration, assuming that T1 0/0/0 is the controller terminating the line from your carrier. In your case, it would be T1 0/2/0 or 0/2/1. You can also look at "show controller t1 [interface]" (e.g. show controller t1 0/2/0) and pay particular attention to "Line Code Violations", "Slip Secs", and "Errored Secs" counters. Note that the "Clock Source Line" status in the "show controller t1" command is not enough to verify you have proper clocking.
Example of a "bad" clock scenario:
SiteA-RTR(config-voiceport)#do sh controller t1 0/0/0
T1 0/0/0 is up.
Applique type is Channelized T1
Cablelength is long 0db
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20080918, FPGA: 13, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (483 seconds elapsed):
3 Line Code Violations, 4 Path Code Violations
207 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
206 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 109 Unavail Secs
To make sure the clock is set correctly, look at the "network-clock-participate" and the "network-clock-select" commands.
For the ISDN status information, you can use "show isdn status". For example:
SiteA-RTR(config)#do sh isdn stat
Global ISDN Switchtype = primary-ni
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x800000FF
Number of L2 Discards = 0, L2 Session ID = 15
*Note: The above is output from a gateway provisioned to use MGCP.
For all gateways, just make sure that Layer 1 is "ACTIVE" and that Layer 2 has "MULTIPLE_FRAME_ESTABLISHED".
You can also use the "show isdn service" command to look at status for individual B-channels.
SiteA-Rtr#sh isdn service
PRI Channel Statistics:
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not appl
ISDN Se0/0/0:23, Channel [1-24]
Configured Isdn Interface (dsl) 0
Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
Use the "Channel State" and "Service State" to verify things are what you expect.
HTH.
-Bill (@ucguerrilla)
http://ucguerrilla.com

Obtaining the number of voice channels with SNMP

Is it possible to get the number of channels in use across a number of voice gateway via SNMP from CallManager?
In RTMT I view the see "MGCP PRI: Channel Active" statistic within Gateway Activity. This is a figure for the whole CCM cluster. I've looked at the CISCO-CCM-MIB but cannot find a reference to this particular value. Can this value be retrieved through the use of SNMP?

it is possible to retrieve the number of channels in use from SNMP.
use PERFMON >
PerformanceObject = MGCP Gateways
Counter = PRIChannelsActive
Instances = All (or which ever you need to view)
this information is also available in RTMT as well as most other SNMP packages such as NetIQ

Adding voice commend to dial number in address book

Id like to figure out how to be able to add a voice commend that would enable me to dial phone number from my address book when i have my cellphone linked with my MBP via bluetooth.
Something along the lines of "Make call to FIRST_NAME LAST_NAME at (WORK, HOME, MOBILE)"
anyone have any ideas

There is a collection of AppleScripts called Mail Scripts you can download at this link.
http://homepage.mac.com/aamann/Mail_Scripts.html
The script you want to use from the collection of included scripts is Add Addresses (Mail) which adds addresses found in the selected messages (in the header fields "From", "To", "Cc", and "Bcc") to the Address Book. This is much more flexible than the "Add Sender to Address Book" available in Mail and provides a convenient way for creating mailing lists.
* Select all messages you want to use for creating the mailing list (for creating a mailing list, you would choose all messages in a mailbox which contains all messages on a certain topic).
* Run the script and you get a list with all addresses contained in the selected messages - mark all the addresses you want to add to the Address Book using the checkboxes.
* From the popup menu, choose which group you want to update with the selected addresses or select "Create new Group..." and enter the name for the new group to be created.
* The script checks whether an address is already present before adding it to the Address Book in order to avoid duplicate entries. If additional information is available for an entry already present in the Address Book (e.g., full name information for an entry which had prior been email-only), the entry will be updated.

Problems adding virtual channels in Max

Has anybody had any problems adding new virtual channels to an existing DAQ OPC configuration? We have a working configuration using 99 tags and adding a further tag to bring the total to 100 causes an OPC server failure. Any help or clues would be appreciated.

Hi mak90,
There are no ways to change the sensor range and physical ranges
programmatically in Traditional DAQ. Under DAQmx, you have Scaling VIs,
Calibration VIs, Task VIs that allow you to change various settings in
MAX programmatically.
Please let us know if you have any further questions.
Best regards,
Nathan Yang
Applications Engineer
National Instruments

Adding HD Channels

I'd like to see both Ovation and TCM added in HD.

I agree. I would also like to add that these HD channels be included, as well: G4 Channel, Cooking Channel, Chiller, TV Land, TV One. Actually, why not provide ALL channels that have a HD broadcast? As much money as we're paying, it sucks to watch low grade channels in anything less than HD when it's available. I switched to Verizon to get away from Comcast and Cablevision, but it looks like I gave up much more.

Adding Auxilary Channels

Hi all, i was just wondering if anyone knew how i could add more auxilary channel strips on my mixer in logic pro 7.2?
I would like to add more than the 2 i already have so that i can route each of the sounds from the ultrabeat drum synth to its own channel strip so i can manipulate them some more by adding effects like delay etc to each sound.
Any help would be much appreaciated thanks,

Hey all
A real quick way to add anything in the Mixer window is to select the "global" setting in the mixer, this will open all objects that are in the environment, then just double click on the objects you want in the mixer and they will appear in the Arrange window and the mixer window. For more info check out page 309 of the manual
Don

Problem in adding voice over presentation

Ok so I want to do a voice over narration on my keynote presentation
I go through the steps of adding audio to my presentation, on the recording screen
the red button is not blinking and my voice is not recording onto the presentation can anyone help pls
Ogay

This is my recommendation.
Step 1) Create your Keynote presentation.
Step 2) Record your speaking part (separately) with an audio recording program.
I recommend Audacity. It's a great free application. You can download Audacity from www.sorceforge.com among other places. You'll also want to get the Audacity LIB file that is needed to export your audio recording to .mp3.
3) From the Keynote Inspector, under the Document tab, go to the audio section and add your new voice recording file to the Soundtrack.
4) From Keynote, go to File, Record Slideshow.
While the audio plays, click through your slides. This is getting the slide change time down.
5) From Keynote, go to File, Export. Make sure to have the following settings:
Playback Uses: Recorded Timing
Include audio (sound files, movie audio): NOT SELECTED
Include the slideshow soundtrack: NOT SELECTED
Include the slideshow recording: SELECTED!!!
Click next and your Keynote presentation should export to a .mov file (keep in mind, this .mov file has no audio. we'll take care of that in the last steps).
6)Open the new .mov Keynote presentation in Quicktime Pro.
7) Open the original .mp3 sound file in Quicktime Pro.
8) Select All of the .mp3 file and then click CTRL + C to copy it.
9) Go back to the .mov Keynote file and Select All. Then go up to Edit > Add to Movie.
This adds your sound and movie file together.
10) Finally click on File > Save As. Select, Save As A Self-Contained Movie and you're done.
I Hope This Helps!

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