You are on page 1of 2

Cisco VG224 con Asterisk

Comparto la configuración de un cisco vg224 usando el protocolo SIP para comunicars
e con un servidor asterisk, no se si con MGCP se puede tener una mayor integraci
on, agradecere mayor informacion si alguien ya realizo alguna configuracion de e
sta manera Saludos, version 12.4 no service pad service timestamps debug datetim
e msec localtime show-timezone service timestamps log datetime msec localtime sh
ow-timezone no service password-encryption ! hostname V 224 ! boot-start-mar!er
boot-end-mar!er ! logging message-counter syslog logging bu""ered 1#$$$ enable p
assword cisco ! no aaa new-model cloc! timezone %& -' ! ! ip source-route ip ce"
! ! ! ! ! ! voice call send-alert voice rtp send-recv ! voice service pots ! vo
ice service voip sip bind control source-inter"ace (ast)thernet$*$ bind media so
urce-inter"ace (ast)thernet$*$ ! voice class codec 1 codec pre"erence 1 g+11ulaw
!
1#0.1 ===> IP del mi router ! ip http server no ip http secure-server .$.2$2.2$2
.aiting &-.$.$.$ 1/2.11 2''.$ ===> IP del VG224 duple2 auto speed auto ! inter"a
ce (ast)thernet$*1 no ip address shutdown duple2 auto speed auto ! ip "orward-pr
otocol nd ip route $.1#0.$.! ! ! ! ! ! ! ! ! ! ! ! voice-card $ ! ! application
service dsapp param call.2''.$ $.2''.) ! global service de"ault dsapp ! ! ! ! !
archive log con"ig hide!eys ! ! ! ! ! inter"ace (ast)thernet$*$ ip address 1/2.
! ! ! control-plane ! ! ! voice-port 2*$ timeouts call-disconnect ' station-id n
ame 3harly station-id number 1/0$ caller-id enable ! voice-port 2*1 timeouts cal
l-disconnect ' station-id name 4ito station-id number 1/01 caller-id enable ! vo
ice-port 2*2 ! voice-port 2*1 ! voice-port 2*4 ! voice-port 2*' ! voice-port 2*#
! voice-port 2*+ ! voice-port 2*0 ! voice-port 2*/ ! voice-port 2*1$ ! voice-po
rt 2*11 ! voice-port 2*12 ! voice-port 2*11 ! voice-port 2*14 ! voice-port 2*1'
! voice-port 2*1# ! voice-port 2*1+ .
! voice-port 2*10 ! voice-port 2*1/ ! voice-port 2*2$ ! voice-port 2*21 ! voice-
port 2*22 ! voice-port 2*21 ! ! ! dial-peer voice 1$ pots destination-pattern 1/
0$ port 2*$ authentication username 1/0$ password 1214' ===> Extensión 1980 cre d e
n mi sterisk ! dial-peer voice 11 pots destination-pattern 1/01 port 2*1 authent
ication username 1/01 password 1214' ===> Extensión 1981 cre d en mi sterisk ! dial
-peer voice 12 pots port 2*2 ! dial-peer voice 11 pots port 2*1 ! dial-peer voic
e 14 pots port 2*4 ! dial-peer voice 1' pots port 2*' ! dial-peer voice 1# pots
port 2*# ! dial-peer voice 1+ pots port 2*+ ! dial-peer voice 10 pots port 2*0 !
dial-peer voice 1/ pots port 2*/ .
! dial-peer voice 2$ pots port 2*1$ ! dial-peer voice 21 pots port 2*11 ! dial-p
eer voice 22 pots port 2*12 ! dial-peer voice 21 pots port 2*11 ! dial-peer voic
e 24 pots port 2*14 ! dial-peer voice 2' pots port 2*1' ! dial-peer voice 2# pot
s port 2*1# ! dial-peer voice 2+ pots port 2*1+ ! dial-peer voice 20 pots port 2
*10 ! dial-peer voice 2/ pots port 2*1/ ! dial-peer voice 1$ pots port 2*2$ ! di
al-peer voice 11 pots port 2*21 ! dial-peer voice 12 pots port 2*22 ! dial-peer
voice 11 pots port 2*21 ! dial-peer voice 1$$ voip destination-pattern 1555 voic
e-class codec 1 session protocol sipv2 session target sip-server dtm"-relay sip-
noti"y rtp-nte no vad .
version 12.' ==> IP !er"er Asterisk no transport tcp o""er call-hold conn-addr !
----------------------------------------------Comparto la configuración de un cisc
o vg224 usando el protocolo SIP para comunicarse con un servidor asterisk.4 no s
ervice pad service timestamps debug datetime msec localtime show-timezone servic
e timestamps log datetime msec localtime show-timezone no service password-encry
ption ! hostname V 224 ! boot-start-mar!er boot-end-mar!er ! logging message-cou
nter syslog logging bu""ered 1#$$$ enable password cisco ! no aaa new-model cloc
! timezone %& -' ! ! ip source-route ip ce" ! ! ! ! ! ! voice call send-alert .
agradecere mayor informacion si alguien ya realizo alguna configuracion de esta
manera Saludos. no se si con MGCP se puede tener una mayor integracion.! sip-ua
sip-server ipv461/2.2$2.1#0.
2''.11 2''.aiting &-.) ! global service de"ault dsapp ! ! ! ! ! archive log con"
ig hide!eys ! ! ! ! ! inter"ace (ast)thernet$*$ ip address 1/2.$ duple2 auto spe
ed auto ! inter"ace (ast)thernet$*1 no ip address ===> IP del VG224 .1#0.2''.2$2
.voice rtp send-recv ! voice service pots ! voice service voip ! voice class cod
ec 1 codec pre"erence 1 g+11ulaw ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card $ ! ! appl
ication service dsapp param call.
$.$.1#0.2$2.$ 1/2.shutdown duple2 auto speed auto ! ip "orward-protocol nd ip ro
ute $.$.$ $.1 ! ip http server no ip http secure-server ! ! ! control-plane ! !
! voice-port 2*$ mwi timeouts call-disconnect ' station-id name 3harly station-i
d number 1/0$ caller-id enable ! voice-port 2*1 mwi timeouts call-disconnect ' s
tation-id name 4ito station-id number 1/01 caller-id enable ! voice-port 2*2 ! v
oice-port 2*1 ! voice-port 2*4 ! voice-port 2*' ! voice-port 2*# ! voice-port 2*
+ ! voice-port 2*0 ! voice-port 2*/ ! voice-port 2*1$ ! voice-port 2*11 ! ===> I
P del mi router .$.
voice-port 2*12 ! voice-port 2*11 ! voice-port 2*14 ! voice-port 2*1' ! voice-po
rt 2*1# ! voice-port 2*1+ ! voice-port 2*10 ! voice-port 2*1/ ! voice-port 2*2$
! voice-port 2*21 ! voice-port 2*22 ! voice-port 2*21 ! ! ! dial-peer voice 1$ p
ots destination-pattern 1/0$ port 2*$ authentication username 1/0$ password 1214
' ===> Extensión 1980 cre d en mi sterisk ! dial-peer voice 11 pots destination-pat
tern 1/01 port 2*1 authentication username 1/01 password 1214' ===> Extensión 1981
cre d en mi sterisk ! dial-peer voice 12 pots port 2*2 ! dial-peer voice 11 pots
port 2*1 ! dial-peer voice 14 pots port 2*4 ! dial-peer voice 1' pots port 2*'
! .
dial-peer voice 1# pots port 2*# ! dial-peer voice 1+ pots port 2*+ ! dial-peer
voice 10 pots port 2*0 ! dial-peer voice 1/ pots port 2*/ ! dial-peer voice 2$ p
ots port 2*1$ ! dial-peer voice 21 pots port 2*11 ! dial-peer voice 22 pots port
2*12 ! dial-peer voice 21 pots port 2*11 ! dial-peer voice 24 pots port 2*14 !
dial-peer voice 2' pots port 2*1' ! dial-peer voice 2# pots port 2*1# ! dial-pee
r voice 2+ pots port 2*1+ ! dial-peer voice 20 pots port 2*10 ! dial-peer voice
2/ pots port 2*1/ ! dial-peer voice 1$ pots port 2*2$ ! dial-peer voice 11 pots
port 2*21 ! dial-peer voice 12 pots port 2*22 .
2$2.! dial-peer voice 11 pots port 2*21 ! dial-peer voice 1$$ voip destination-p
attern 1555 voice-class codec 1 session protocol sipv2 session target sip-server
dtm"-relay sip-noti"y rtp-nte no vad ! sip-ua authentication username cisco pas
sword cisco121 retry invite 1 retry response 1 retry bye 1 retry cancel 1 timers
trying 1$$$ mwi-server ipv461/2.11 insecure:very .11 .1#0.' ==> IP !er"er Aster
isk ! 7hora creamos un &run! S89 en nuestro asteris! PEE# allow:ulaw canreinvite
:no conte2t:"rom-internal disallow:all dtm"mode:r"c2011 host:1/2.'6'$#$ e2pires
1#$$ sip-server ipv461/2.uali"y:yes secret:cisco121 type:peer username:cisco $!E
# allow:ulaw canreinvite:no conte2t:"rom-internal disallow:all dtm"mode:r"c2011
host:1/2.2$2.2$2.1#0.2$2.1#0.2$2.' e2pires 1#$$ port '$#$ transport udp registra
r ipv461/2.1#0.1#0.
uali"y:yes type:user n el !G224 verificamos "ue los ane#os creados esten registr
ados en nuestro asterisk$ V 224<show sip register status =ine peer e2pires>sec?
registered :::::::::::: ::::::::::::: :::::::::::: ::::::::::: 1/0$ 1$ 11 yes 1/
01 11 11 yes ..