Testing

How to Build an Emulated CCIE Voice Lab

noreply@blogger.com (Brad) — Wed, 07 Oct 2009 14:46:00 +0000

I found this link on http://www.techexams.net/forums. I'd heard of BlindHog before, but didn't know he had a guide for emulating a CCIE Voice lab! Apparently he piggybacked (pun definitely intended) off of someone else's work, but he added on top of it with the guides and videos on making it all work.

Wow, this is awesome. Great work, guys.

http://www.blindhog.net/how-to-build-a-voice-lab-with-gns3-and-vmware/

School = Blog Break Time

noreply@blogger.com (Brad) — Tue, 22 Sep 2009 20:33:00 +0000

So I'm back in school and finding time to work 40 hours, go to class, study, train my dog (duck hunting), and mix in a bit of sleeping/eating has proven difficult! This blog drew the short end of the stick, so it will be put on hold until I can get a random bit of time to keep up my VOIP studying.

/sadface

Wireless AP Scenario Lab Complete

noreply@blogger.com (Brad) — Tue, 01 Sep 2009 18:54:00 +0000

I decided to take some time off of CBT's and reading to get in the thick of things with some labs.

I completed the scenario-based lab on setting up the embedded wireless access point and a wireless IP phone. The devices I used were:

Cisco UC520

Cisco 7920 Wireless IP Phone

The setup was actually quite easy using the Cisco Configuration Assistant. I already had the community built for my lab, so all that was required to complete this lab was to:

1.) Set up the wireless access point for voice usage

2.) Ensure the voice AP uses the voice vlan.

3.) Set up the network profile in the wireless IP phone.

4.) Connect and test voice calls using the voice AP.

After completing these steps, I had another person test calls with me. This was tested in an environment with a large amount of wireless saturation. It worked great and I could get about 50-60 yards before the signal would begin degrading. Another 10-20 yards before the call would drop.

CBT Status 9

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 19:14:00 +0000

Now watching:

Gateways and Trunks: Connecting CME to Other Voice Systems

RTP and RTCP

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 19:08:00 +0000

Real-time Transport Protocol (RTP) - transports audio between devices

Uses random, even UDP ports between 16,384 and 32,766

Real-time Transport Control Protocol (RTCP) - transports call statistics between devices

Uses random, odd UDP ports between 16,385 and 32,767.

Digital Signal Processors

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 19:05:00 +0000

Digital Signal Processors (DSPs) offload media processing functions from voice processing equipment.

They provide:

Coding - applying the codec and compressing the audio

Transcoding - if codec changes are needed, such as for a phone that can only receive G.729, the DSP will provide it

Media Termination Point (MTP) - does the work behind the hold feature

Conferencing - provides conferencing by combining audio from various sources into one audio stream

Bandwidth Savings Measures

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 19:00:00 +0000

Voice Activity Detection (VAD): Why waste bandwidth sending silence? VAD removes this need and saves an average of 35% bandwidth.

Compressed RTP: Compresses network and transport layer headers from 40 bytes to 2-4 bytes. Bandwidth savings are codec dependent because each with have a different overall total size, but it's around 40% with G.729, the most common Cisco codec.

Common Audio Codecs

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 18:33:00 +0000

Codec	Bandwidth/Call	Quality
G.711	64Kbps	MOS = 4.1
G.729	8Kbps	MOS = 3.92
G.729A	8 Kbps	MOS = 3.7
G.726	32 Kbps	MOS = 3.85
G.728	16 Kbps	MOS = 3.61
Internet Low Bitrate Codec (ILBC)	15.2 Kbps	MOS = 4.1

G.729 is the most popular codec among Cisco devices. G.729A uses less Digital Signal Processors, so less hardware required per device.

The industry brought out ILBC as an open-source, industry standard codec that was able to compress voice while having the same quality as uncompressed voice. This is only supported on newer phones and devices.

MOS stands for "Mean Opinion Score" and is a voice quality rating scale from 1(worst) to 5(best). Normal uncompressed PSTN voice quality is 4.1, which is the same as G.711 and ILBC, except ILBC requires the least amount of bandwidth. ILBC calls requires less than 25% of the bandwidth required by Normal PSTN and G.711 voice.

Converting Voice to Bits

noreply@blogger.com (Brad) — Tue, 25 Aug 2009 18:21:00 +0000

Steps to convert Voice into Bits for VoIP:

1.) Sample the signal

2.) Quantize each sample (give it a number to represent the sample)

3.) Convert the number the binary (to computer language!)

4.) Optionally, compress the signal

CBT Status 8

noreply@blogger.com (Brad) — Thu, 20 Aug 2009 19:54:00 +0000

Now watching:

Gateways and Trunks: Understanding Voice CODECs

CME GUI

noreply@blogger.com (Brad) — Wed, 19 Aug 2009 19:44:00 +0000

The CME GUI allows you to manage some CME features via a GUI.

To configure the CME GUI:

Conf t

Ip http secure-server (or just server)

Ip http path flash:/gui

Ip http authentication local

Telephony-service

Web admin system name VOIPNewb secret 0 SECRETPASS

Dn-webedit (allows changing directory numbers in webpage)

Time-webedit (allows changing system time in webpage)

To reach the CME GUI:

Http(s)://ROUTERIP/ccme.html

CME Music on Hold

noreply@blogger.com (Brad) — Wed, 19 Aug 2009 19:34:00 +0000

Music on Hold streams a music file (.wav or .av) in flash memory to a caller while they're on hold.

To copy music from tftp server to flash:

Copy tftp flash

Source filename []? MyMusic.wav

Destination filename [MyMusic.wav]?

To configure Music on Hold:

Conf t

Telephony-service

Moh MyMusic.wav

Multicast moh 239.1.1.55 port 2100

CME After Hours Call Blocking

noreply@blogger.com (Brad) — Wed, 19 Aug 2009 19:27:00 +0000

After Hours Call Blocking is a feature that does just what it says… it blocks calls after hours. Note that these are OUTGOING calls. Example: Your business may close at 5PM and an employee stays late to use your phone and talk to his mother long-distance. With after hours call blocking, you could block any calls or you could just block long-distance calls to discourage this activity.

To configure after hours call blocking:

Conf t

Telephony-service

After-hours day mon 17:00 8:00

After-hours day tue 17:00 8:00

After-hours day wed 17:00 8:00

After-hours day thu 17:00 8:00

After-hours day fri 17:00 8:00

After-hours date dec 25 00:00 00:00

After-hours block 1 91………. (blocks any (1-XXX-XXX-XXXX number)

After-hours block 2 91900……. (blocks any 1-900 number)

To exempt a phone from after hours call blocking:

Conf t

Ephone 1

After-hour exempt

Exit

Ephone 2

Pin 1234 (assigns a PIN so the person can enter the code and then be able to make outgoing calls that would previously be blocked by an after hours call blocking pattern)

exit

To activate the above PIN option, logins need to be enabled:

Conf t

Telephony-service

Login timeout 120 clear 23:00 (allows logging in using the pin above, lets them stay logged in for 120 minutes, and clears the login at 23:00. Does not hang up the call if they're on the phone at 23:00, but will clear the login at 23:00, requiring a new login)

To block calls 24/7:

Conf t

Telephony-service

After-hours block pattern 3 91900……. 7-24

CME Paging

noreply@blogger.com (Brad) — Wed, 19 Aug 2009 19:03:00 +0000

Paging allows one phone to communicate with other phones via one-way communication. This means you can pick up your phone, hit a button, and instantly say a message for other phones to hear. An example: Your accounting department has a package they need to sign for, so you page the accounting department group's paging number, tell them that someone needs to come sign for the package, then hang up.

Paging works by adding phones to paging groups. A phone cannot belong to more than one paging group at a time, but a paging group can belong to another paging group.

Paging can be configured through unicast or multicast. Obviously, multicast is more efficient, but not all networks support it.

To configure Multicast Paging:

Conf t

Ephone-dn 32

Number 5510

Paging

Paging ip 239.1.1.100 port 2000

Exit

Ephone 1

Paging-dn 32

Exit

Ephone 3

Paging-dn 32

Exit

Ephone-dn 33

Number 5511

Paging

Exit

Ephone 2

Paging-dn 33

Exit

Ephone 4

Paging-dn 33

Exit

Ephone-dn 34

Number 5512

paging

Paging group 32,33

CME Intercom

noreply@blogger.com (Brad) — Tue, 18 Aug 2009 20:52:00 +0000

Sets up a direct link between an assistant phone and a manager phone, where the assistant phone can press a button which will dial the manager phone and it will automatically answer on speakerphone right away. When it's auto-answered, it is muted, so nothing can be overheard until the mute button is hit to unmute it.

To configure Intercom:

Conf t

Ephone-dn 30

Number A100 (A100 used so it can't be accidently dialed in network)

Intercom A101 label "Manager"

Exit

Ephone-dn 31

Number A101

Intercom A100 label "Assistant"

Exit

Ephone 3

Button 4:30

restart

Exit

Ephone 5

Button 2:31

restart

Exit

CBT Status 7

noreply@blogger.com (Brad) — Tue, 18 Aug 2009 20:39:00 +0000

Currently watching:

Cisco CME: Voice Productivity Features, Part 2

CME Call Pickup

noreply@blogger.com (Brad) — Tue, 18 Aug 2009 20:38:00 +0000

Call Pickup allows a person to answer another ringing phone in the network.

Three different types of Call Pickup:

Directed Pickup - Phones can hit "Pick Up" and dial the number of the phone that's ringing

Local Group Pickup - Phones in same group can hit "More" and hit "GPickUp"

Other Group Pickup - Phone in group 101 can hit "More" and then "GPickUp", then dial other group number to pick up the call ringing in their group

To configure Call Pickup:

Conf t

Ephone-dn 3

Pickup-group 1101

Ephone-dn 4

Pickup-group 1101

Etc.,etc.

If any two phones are ringing at once, hitting "GPickUp" will answer the call that began ringing first.

Answer Parked Calls

noreply@blogger.com (Brad) — Tue, 18 Aug 2009 20:28:00 +0000

After calls are parked, someone ideally should be answering it. Here's how to do that:

1.) Dial the number of the parked call extension

2.) Hit the "Pick Up" key and dial the parked call extension

3.) Hit the "Pick Up" key and dial the * key. This connects only the person that parked the call and only connects them to the person they most recently parked.

CME Call Park

noreply@blogger.com (Brad) — Tue, 18 Aug 2009 20:27:00 +0000

Configuring Call Park:

Conf t

Ephone-dn 30

Number 3001

Park-slot

Exit

Call park does just what it sounds like, it parks a call at an extension. When someone wants to talk with that person, they would dial 3001 to speak to them. For example, "Johnny, call on line 1, Johnny, line 1!"

Call Park Options:

Reserved-for - only the specified extension can use the call slot.

Timeout - after the specified timeout time, if no one answers the parked call, it will hang up on the caller. (Option extension "limit" sets the number of times the person that parked the call will be beeped every X timeout seconds before the caller will be hung up on)

Notify - sends a reminder ring to the specified extension in addition to the reminder ring that is sent to the phone that parked the call.

Only - sends a reminder ring to the extension specified in the notify option and not to the extension that actually parked the call

Recall - returns the call to the phone that parked it after the timeout expires

Transfer - returns the call to the specified extension after the timeout expires

Alternate - returns the call to this second target extension if the recall or transfer target extension is in use on any of its extensions (ringing or connected)

Retry - sets the delay before another attempt to recall or transfer a parked call, in seconds

CME Call Transfers

noreply@blogger.com (Brad) — Mon, 17 Aug 2009 20:40:00 +0000

Configuring Call Transfers:

Conf t

Telephony-service

Transfer-system {full-blind|full-consult|local-consult}

Call Transfer Options:

Full-blind - transfers the call immediately without any warning (appears as a normal incoming call to thereceiving employee)

Full-consult - transfers the call, but places the original caller on hold, while the employee transferring the call waits for the employee receiving the transfer to answer. Once they answer, they have a chance to speak until the employee transferring hits the transfer button to fully complete the transfer. This gives the employee receiving the call advanced warning of who's calling and any necessary time to prepare for the call.

Local-consult - same as full consult, just uses an older cisco proprietary protocol to prevent hairpinning

Configuring Call Transfer Patterns

Conf t

Telephony-service

Transfer-pattern xxxx

The call transfer pattern specifies a number of digits that calls can be transferred to. Much like the forward-pattern or max-length option available with call forwarding, transfer-pattern allows you to restrict transfers to a number of digits, which would usually be used to restrict transferring to the length of digits used internally, so calls cannot be transferred outside.

CME Call Forwarding Options

noreply@blogger.com (Brad) — Mon, 17 Aug 2009 20:11:00 +0000

In the previous post, I included several options without explaining what they do. Some are easy to figure out, others not so easy. Here's an explanation of each:

Call-forward {all|busy|max-length|night-service|noan}

All - forward all calls

Busy - forward calls only when the call would normally receive a busy tone

Max-length - set the max number of digits allowed to be used for call forwarding number specifications

(I'll explain more about this one later in this post)

Night-service - forward calls only when night service is activated

Noan - forward calls only when the call is not answered (when it would normally go to voicemail)

Now, the max-length option might seem like an unnecessary option upon first reading it. However, consider "toll fraud." Employees set their phones to forward to 1-900 numbers or to long distance/international numbers that would normally cost extra. After leaving work, they call their work phone, which forwards the call to the 1-900 or international number they originally set. At that point, the business is paying for the extra charges. Putting a "max-length 4" or however many digits you use internally will keep calls inside your organization. Fully-trusted employees can have their max-length extended.

CME Call Forwarding

noreply@blogger.com (Brad) — Thu, 06 Aug 2009 21:31:00 +0000

Call forwarding allows forwarding of calls to a different phone than it's normally set up to ring on. This can be accomplished on the phone, itself, or within the CME router. Setting call forwarding on the phone only requires pressing the CallFWD button and entering the number to forward calls to.

Configuring Call Forwarding on the CME Router requires:

Conf t

Ephone-dn 2

Call-forward {all|busy|max-length|night-service|noan} 2121 timeout 25

To disable Call Forwarding:

Conf t

Ephone-dn 2

Call-forward max-length 0

CME Directory Configuration

noreply@blogger.com (Brad) — Thu, 06 Aug 2009 21:20:00 +0000

The directory is accessed through IP Phones to find someone's number and contact them.

To configure a directory entry:

Conf t

Telephony-service

Directory entry 1 1010 name "Fax Machine"

Exit

To configure a directory entry for an ephone-dn:

Conf t

Ephone-dn 1

Name John Doe

Exit

That's it. The direct entry into the directory puts in a custom entry. Adding the directory entry to the ephone-dn enters it into the directory, but changes the entry if the number assigned to the ephone-dn changes.

CBT Status 6

noreply@blogger.com (Brad) — Thu, 06 Aug 2009 21:14:00 +0000

Currently watching:

Cisco CME: Voice Productivity Features

CUCME Command Reference

noreply@blogger.com (Brad) — Thu, 06 Aug 2009 19:18:00 +0000

This section on Cisco's site has the complete command reference for Cisco Unified Communications Manager Express:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_cr.html

The complete book can be downloaded on the page linked above, or you can click through the pages sorted alphabetically on the left.

This is an invaluable resource if you're working with CUCME.