Piccolo aggiornamento sul post precedente, ho dovuto fare una piccola correzione alla configurazione postata in precedenza, in quanto non era necessario dtmf-dialing, ed il sintomo che dava era curioso, ovvero il numero di telefono digitato da alcuni apparecchi collegati al PBX classico, erano “storpiati”.
Ad esempio se io digitavo 0577123456 questo mi diventava 0505777711223345455656 ad esempio!!!
Togliendolo tutto ok, inoltre ho creato un nuovo utente per usarlo come “friend” lato pbx ed inoltre ho aggiunto delle rotte per chiamare i cellulari, 8XX e 7XX
#—————————————————————-#
# #
# SN4638/5BIS/UI #
# R3.20 2006-11-17 H323 SIP BRI #
# 2007-07-27T06:39:21 #
# Generated configuration file #
# #
#—————————————————————-#
cli version 3.20
dns-client server 88.149.128.12
webserver port 80 language en
snmp community public ro
system
ic voice 0
profile ppp default
profile call-progress-tone IT_Dialtone
play 1 200 425 -12
pause 2 200
play 3 600 425 -12
pause 4 1000
profile call-progress-tone IT_Alertingtone
play 1 1000 425 -12
pause 2 4000
profile call-progress-tone IT_Busytone
play 1 500 425 -12
pause 2 500
profile tone-set default
profile tone-set IT
map call-progress-tone dial-tone IT_Dialtone
map call-progress-tone ringback-tone IT_Alertingtone
map call-progress-tone busy-tone IT_Busytone
map call-progress-tone release-tone IT_Busytone
map call-progress-tone congestion-tone IT_Busytone
profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
profile voip ASTERISK
codec 1 g729 rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g711alaw64k rx-length 20 tx-length 20
profile pstn default
profile sip default
profile sip asterisk
profile aaa default
method 1 local
method 2 none
context ip router
interface IF_IP_WAN
ipaddress 192.168.1.101 255.255.255.0
no napt-inside
interface IF_IP_LAN
ipaddress 192.168.1.1 255.255.255.0
no napt-inside
context cs switch
digit-collection timeout 4
national-prefix 0
international-prefix 00
routing-table called-e164 RT_2_ISDN
route .%T dest-service HUNTING MT_ITC
routing-table called-e164 RT_ISDN_2_SIP
route 99[1-9].T3 dest-interface IF_S0_PSTN
route 1[1-9].T3 dest-service SER_HUNT_OUT
route 0[1-9].T3 dest-service SER_HUNT_OUT
route 00[1-9].T3 dest-service SER_HUNT_OUT
route 8[0-9].T3 dest-service SER_HUNT_OUT
route 7[0-9].T3 dest-service SER_HUNT_OUT
route 3[0-9].T3 dest-service SER_HUNT_OUT
route default dest-service SER_HUNT_OUT
mapping-table itc to itc MT_ITC
map default to speech
interface isdn IF_S0_PSTN
route call dest-interface IF_S0_PHONE
dtmf-dialing
interface isdn IF_S0_PHONE1
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT
interface isdn IF_S0_PHONE2
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT
interface isdn IF_S0_PHONE3
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT
interface isdn IF_S0_PHONE4
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT
interface sip IF_SIP_ASTERISK
bind gateway GW-ASTERISK
service default
route call dest-table RT_2_ISDN
early-disconnect
remote-party-id called-party
remote-party-id calling-party
address-translation outgoing-call from-header user-part fix 210 host-part call
use profile voip ASTERISK
service hunt-group SER_HUNT_OUT
timeout 6
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
route call 1 dest-interface IF_SIP_ASTERISK
route call 2 dest-interface IF_S0_PSTN
service hunt-group HUNTING
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
drop-cause user-busy
route call 1 dest-interface IF_S0_PHONE1
route call 2 dest-interface IF_S0_PHONE2
route call 3 dest-interface IF_S0_PHONE3
route call 4 dest-interface IF_S0_PHONE4
context cs switch
no shutdown
gateway sip GW-ASTERISK
bind interface IF_IP_WAN router
service default
domain 192.168.1.00
realm 192.168.1.100
authentication 210 password 210
default-server 192.168.1.100 loose-router
registrar 192.168.1.100 5060
user 210
session-timer 1800
gateway sip GW-ASTERISK
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown
port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown
port bri 0 0
clock auto
encapsulation q921
power-feed
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE1 switch
port bri 0 0
no shutdown
port bri 0 1
clock auto
encapsulation q921
power-feed
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE2 switch
port bri 0 1
no shutdown
port bri 0 2
clock auto
encapsulation q921
power-feed
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE3 switch
port bri 0 2
no shutdown
port bri 0 3
clock auto
encapsulation q921
power-feed
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE4 switch
port bri 0 3
no shutdown
port bri 0 4
clock auto
encapsulation q921
q921
protocol pmp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF_S0_PSTN switch
port bri 0 4
no shutdown
Lo username 210 in sip.conf
[210]
username=210
type=friend
secret=210
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=210@device
host=dynamic
dtmfmode=inband
context=from-internal
canreinvite=no
disallow=all
allow=alaw
)
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