CISCO VOIP | اساسيات الـ VOICE OVER IP | فكره عامة عن الـ VOIP

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شرح VOIP
اساسيات VOIP
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VoIP and SIP What are they?

What is VoIP? What is SIP? How are they the same? How are they Different?

In this video I explain what each one is and how they are used in real life.

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How to Configure an FXO Port to Ring VoIP Phone on CUBE and CUCM

How to Configure an FXO Port to Ring VoIP Phone on CUBE and CUCM
FXO Voice Port Configuration
• Unlike a T1 Card there is no configuration needed after FXO Card installation
• Each FXO port will need to be configured and there are several parameters that need to be setup.
• There are other configuration setting that may be needed depending on your setup and location; I have just listed the basic settings I used to setup our Ecuador office.
o connection plar (opx)
 This command will ring an extension from CUCM
o Signal
 Loopstart – Standard residential phone lines are configured as FXS loop-start. Loop-start is the default signaling on Cisco IOS FXS and FXO voice ports. In order to change it, issue the signal ground-start voice-port command. Reset the voice-port after any changes with the shutdown/no shutdown command sequence. This is the default setting.
• Source –

o Supervisory disconnect dualtone mid-call
 supervisory disconnect – Configures supervisory disconnect signaling on the FXO port. Supervisory disconnect signaling is a power denial from the switch that lasts at least 350 ms. When this condition is detected, the system interprets this as a disconnect indication from the switch and clears the call. You should disable supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch. Typically, supervisory disconnect is available when connecting to the PSTN and is enabled by default. When the connection extends out to a PBX, you should verify the documentation to ensure that supervisory disconnect is supported
o Cptone [Country Code]
 This command defines the detection of call-progress tones generated at the local interface. It does not affect any information passed to the remote end of a connection, and it does not define the detection of tones generated at the remote end of a connection. Use the cptone command to specify a regional analog voice interface-related default tone, ring, and cadence setting for a specified voice port. The current configured cptone setting is visible under the ‘show voice-port [port]’ command
 voice class custom-cptone [WORD]
• Adding to the voice-port
o Voice-port 0/1/2
supervisory custom-cptone [WORD]
• World Tone Database -,Audible_ring_tone
o Dial-type
 DTMF is set as default
o trunk-group
 An interface can belong to only one trunk group. Multiple interfaces can belong to the same trunk group.
 Example
• voice-port 0/1/3
trunk-group FXO
connection plar opx 2001
caller-id enable
 Dial-peer Example
• dial-peer voice 1000 pots
destination-pattern .T
trunkgroup FXO
o Ring Number

Dial-peer Configuration
Because the incoming call is coming from the PBX into the FXO port, the router answers the call. We want the incoming call to ring a VoIP phone that is connected to CUCM, and because we have set connection plar opx 2001 command, the router is going to ‘dial out’, and we need to create a VoIP dial-peer for this.

dial-peer voice 5000 voip
description ** POTS/FXO IN **
destination-pattern 2001
session target ipv4: (This is the IP of CUCM)
dtmf-relay h245-alphanumeric
no vad

dtmf-relay h245-alphanumeric -This is an out-of-band DTMF relay mechanism that transports the DTMF signals using H.245, which is the media control protocol of the H.323 protocol suite.

Source – (

We need to add our other dial-peers, for outbound calls through the FXO port.

dial-peer voice 4000 pots
trunkgroup FXO
description ** TV CABLE POTS OUT LOCAL QUITO #’s **
destination-pattern [2-9]……
forward-digits all

dial-peer voice 4001 pots
trunkgroup FXO
description ** TV CABLE POTS OUT LD/CELL #’s **
destination-pattern [01]……..
forward-digits all

Good Sources

CUCM Configuration Checklist
• Date/Time Group
• Region
• SIP Trunk
• SIP Profile
• Device pool for CUBE
• Device pool for PHONES
• Media Termination Point (MTP)
• Media Resource Group (MRG)
• Media Resource Group List (MRGL)
• Partition
• Calling Search Space (CSS)
• Route Group
• Route List
• Route Pattern
• Create H.323 Gateway


Configuring VoIP Phones in Cisco Packet Tracer

Please read the below command .

DHCP server on RouterA (2811 router) :

ip dhcp pool VOICE
option 150 ip

Configure the Call Manager Express telephony service on RouterA .
max-ephones 5
max-dn 5
ip source-address port 2000
auto assign 1 to 6

Configure the phone directory for IP Phone 1

ephone-dn 1
number 123

ephone 1
device-security-mode none

type 7960
button 1:1