debug voip dialpeer // see numbers being dialed, and calls being placed
show voice call summary//see active calls
show dialpeer voice summary // see a list of extensions next hops IE Go to this router to get this ext.
debug ephone detail // see ephone as it registers
—Phase 1 upgrade.
New router is purchased and your exsisting PABX is used to call over the WAN to brance offices.
—Phase 2 upgrade
running totally VOIP at a site .
There are 2.5 types of VOIP servers.
when setting up a branch office you have 2 options
Use a VPN to connected to the CUCM server, or have a UCE router on site.
Going over VPN if the link goes down, phones will not work. less cost.
Having a UCE on site means less bandwidth used, and calls can still be made if
VPN link goes down.
CCP(cisco configuration professional) is the SDM GUI for VOIP router devices.
—-Unified communications manager. (Large business)
cisco unity presence – Tracking a user state. are they logged in.
-multi site support
-supports 60k IP phones per cluster. Cluster is a single database in an area.
-there is redundancy in each cluster.
—-unified communications business edition.
middle ground 500 IP phones
less features.
no redunacy less UC manager.
Does cisco unity, cisco mobility, and cisco communications manager.
—-Unified communications manager express
(small/ medium business)
Built into the router. 450 phones supported. 100 phones tops before problems.
Uses unity express for voicemail.
UC500 (is really call manager express)
—Voicemail – Unity
Old system, being phased out, ran on server 2000, direct tether to exchange.
—Voicemail – Unity connection
Recently tied to microsoft exchange. 20k accounts per device.
used with CUCM.
—Voicemail – Unity express.
supports 250 accounts, only works for about 100.
ties in with exchange, supports a menu style system (IVR)
auto attendant.
Can be either a ISM(internal service module) (Riser board) or SM(service module) Module
that slides in.
—Cisco presence
Pull information about the user to display (if they are on the phone)
Does instant messaging,
–Analogue phones
at home when you pickup the handset, the circuit is bridged and you get a dialtone.
There are 2 wires ring and tip. This is known as loop start.
The signal needs to be regenerated every so often, leading to loss in quality.
In the office environment this can cause a problem known as GLARE.
you pickup the phone and it attempts to use a line that is just about to be utilized
by the PABX to recieve a call. to overcome this problem we have the PABX sending a signal
to the office requesting to reserve the line for use, this is called ground start.
When ringing uses an AC current instead of a DC current.
2 types of dialing, pulse dialing, which uses a break in the ring an tip to signal which
number to dial.
DTMF – dual tone multifrequency uses 2 frequency sounds to indicate the number dialed.
—Digital phones.
Muliple calls can go over one wire unlike analogue wiring.
There are 2 encoding standards – A-law (logical conversion) and Mu-law (Wildcard inverse conversion)
Routers have the ability to convert between the 2 standards, different countries use different standards.
Converting is called Transcoding.
—Compression
Each call un compressed uses 64kb of link speed. there are 3 types of compression
Send all samples
send the changes – when there is a change in the binary number being sent, send the changes
use a codebook – route saves common known sounds, and uses a reference in the codebook.
G.729 is the compression type that drops the needed link speed to 8kbs.
—Call control modes: Distributed
VOIP Router handles the connection amoungst the phones, phones do the conversion and direct talking once the session is established.
Redunancy, in that if one VOIP router goes down, another can be used.
—Call control modes: Centralised
There is a central CUCM server that handles the connections of the phones.
VOIP routers convert analogue signals via DSP to digital to talk with VOIP systems.
—SRST (survivalable remote site telephony)
If the connection is lost to the centralised CUCM server, use the POTS/ or take this path.
—Signalling protocols
Protocols used to communicate to CUCM server, to advised when a button is pressed.
H.323 //High overhead. Used between voice gateways. Used for a distributed call control modes
MGCP //used between voice gateways. Used for Centeralized call control modes.
SIP //industry standard leader
SCCP //Skinnny cisco propertiary
—Streaming protocols
RTP (Real-time transport protocol) //This is the voice going across the network
RTCP(Real-time transport control protocol) //This is the stats going across the network.
You can press the “i” or help button twice on a VOIP phone while you are on a call, if you want to see stats.
–WAN and VOIP.
VOIP only uses the WAN to make calls between offices. All external calls take the POTS PSTN lines.
Alternatively you can route all your calls to an ITSP (Internet telephone service provider) to direct all your calls over a internet.
No QOS over the internet, usually cheaper.
—DSP’s
convert signals, coming in from the PSTN and going out the PSTN.
Modules inside the VOIP router.
—Remote offices
Remote offices, have 2 options when dialing externally.
Route calls through the WAN/ then through the Office PSTN or
you can have a PSTN line going to the home (great for redundancy if the WAN goes down)
–TEHO(Tail End hop off)
TEHO is the ability to use the WAN to get local calls in a country where you have a branch office.
—distributed Multi-cluster design.
Each site has its own brain and can be linked together.
—Two modes of power
Inline power was developed by cisco, Power of ethernet is the industry standard.
midspan power is when the patch panel delivers power to your device, switches dont need to
supply power to the phone, injects it into the line.
If you are connecting a non cisco phone, the phone will not use CDP to utilize the most effective voltage, it will provision the full amount of watts.
You can set the amount of power provisioned manually.
By default the switch should send power and power all the devices.
There are acouple of settings available under the interface
show power inline //see how much power is being used.
int fa1/1
power inline <auto/delay/shutdown> //auto detect, delay before turning off power, no power sent
—Switch PORT configuration.
//create your vlans x2
conf t
vlan <number> name <name>
//assign the voice and access vlans to your interfaces
int fa1/1
switchport mode access //set port to access for clients,
switchport access vlan 10 //set PC vlan
switchport voice vlan 50 //set Phone Vlan. must be cdp phone?
port should appear under 2 vlans under sh vlan br.
The old way was to configure a trunk to the phone and then use the native vlan(untagged for the PC) This was a security risk.
If your phone/switch is not running cdp then you will need to set the vlan on the phone. through the menu.
—-IP phone boot process
1. recieves power over ethernet
2. Communicates via CDP to get vlan information
3. recieves DHCP request/reply with -150 attribute which says location of TFTP server
4. downloads its configuration file from the TFTP server, config file contains contacts and
location of CUCM or UCME server
5. contacts UCME server.
—SPanning-tree
It is best practice to put your VOICe traffic in lower vlan numbers than your data vlans.
This is because the lower VLANs are converged first when spanning-tree is converging.