Scribd
Upload
56 views4 pages

Configuracion de Telefonos VOIP

The document configures a Cisco router as a SIP client of an Asterisk server to enable VoIP calls between SIP phones, setting up dial peers, directories, phones and their associations, enabling SIP registration and calls between SIP endpoints. It also shows the configuration of extensions in Asterisk to route calls between the SIP URIs configured on the router to the appropriate internal extensions.

Uploaded by

Mundo Root
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
56 views4 pages

Configuracion de Telefonos VOIP

The document configures a Cisco router as a SIP client of an Asterisk server to enable VoIP calls between SIP phones, setting up dial peers, directories, phones and their associations, enabling SIP registration and calls between SIP endpoints. It also shows the configuration of extensions in Asterisk to route calls between the SIP URIs configured on the router to the appropriate internal extensions.

Uploaded by

Mundo Root
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
You are on page 1/ 4

- Dial Peer envía llamadas a Asterisk

dial-peer voice 13 voip


destination-pattern [2-9]...
session protocol sipv2
session target ipv4:192.168.1.100
dtmf-relay rtp-nte
codec g711ulaw

- Configuración router CME directorios y teléfonos con el protocolo SCCP

telephony-service
max-ephones 30
max-dn 100
ip source-address 172.16.1.10 port 2000
voicemail 2105

- configurar directorios (dn1 y dn2)

ephone-dn 1 dual-line
number 1001
label Telecomunicaciones
call-forward busy 2107
call-forward noan 2106 timeout 10

ephone-dn 2 dual-line
number 1002
label Redes
call-forward busy 2107
call-forward noan 2106 timeout 10
- configurar teléfonos 1 y 2 y asociarlos a los directorios 1 y 2 creados

ephone 1
mac-address 0200.4C4F.4F50
button 1:1

ephone 2
mac-address 000C.2947.A416
button 1:2

- habilitar el servicio VoIP y las llamadas entre teléfonos SIP

voice service voip


allow-connections sip to sip
sip
registrar server expires max 1200 min 300

- Router CME como cliente SIP de Asterisk

sip-ua
mwi-server ipv4:192.168.1.100 expires 86400 port 5060 transport tcp unsolicited
registrar ipv4:192.168.1.100 expires 3600 secondary

- Configuración router CME directorios y teléfonos con el protocolo SIP

voice register global


mode cme
source-address 172.16.1.10 port 5060
max-dn 10
max-pool 5
create profile

- configurar directorios (dn1 y dn2)

voice register dn 1
number 1003

voice register dn 2
number 1004

- configurar teléfonos 1 y 2 y asociarlos a los directorios 1 y 2 creados

voice register pool 1


id mac DEAD.BEEF.0001
number 1 dn 1
dtmf-relay sip-notify
username sena password 1234
codec g711ulaw
voice register pool 2
id mac DEAD.BEEF.0001
number 2 dn 2
dtmf-relay sip-notify
username cisco password 1234
codec g711ulaw

sip.conf
[5001]
type=friend
host=dynamic
language=es
context=local
mailbox=1001
allow=ulaw
allow=alaw

[5003]
type=friend
host=dynamic
context=local
language=es
mailbox=1003
allow=ulaw
allow=alaw

[5002]
type=friend
host=dynamic
context=local
language=es
mailbox=1003
allow=ulaw
allow=alaw

[2100]
type=friend
host=dynamic
language=es
context=local
mailbox=2100
allow=ulaw
allow=alaw

extensions.conf

[local]

exten => _30XX,1,Dial(SIP/${EXTEN})


exten => _30XX,n,Hangup()
exten => _500X,1,Dial(SIP/${EXTEN},192.168.0.100)
exten => _500X,n,Voicemail(${EXTEN}@default)
exten => _500X,Hangup

exten => 2105,1,VoicemailMain(${CALLERID(num)}@default)


exten => 2105,n,Hangup

You might also like