PATTON 4120 CHIAMATA ENTRANTE ANONIMA

sartons

Gold Partner
Advanced Certified
Registrato
10 Luglio 2015
Messaggi
2
Buongiorno, ho installato un patton 4120. Tutto funziona benissimo ad eccezione delle chiamate in ingresso,che arrivano, ma che mi arrivano tutte anonime.
Di seguito la configurazione del Patton, che tra l'altro è quella di default del 3cx

#----------------------------------------------------------------#
# #
# SN4120/2BIS4V #
# R6.3 2013-05-01 H323 SIP #
# 2015-07-09T15:52:07 #
# SN/00A0BA0B357E #
# Generated configuration file #
# #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +01:00
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.1.2 port 123 version 4
sntp-client local-clock-offset

system

ic voice 0
low-bitrate-codec g729

system
clock-source 1 bri 0 0
clock-source 2 bri 0 1

profile ppp default

profile call-progress-tone defaultDialtone
play 1 200 425 0
pause 2 200
play 3 600 425 0
pause 4 1000

profile call-progress-tone defaultBusytone
play 1 200 425 -7
pause 2 200

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g729 rx-length 20 tx-length 20
fax transmission 1 relay t38-udp
modem transmission 1 bypass g711alaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_LAN
ipaddress 192.168.1.130 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch

routing-table called-e164 RT_ISDN_TO_SIP_0
route T2 dest-interface IF_SIP_0 MAPPING_INCOMING_CALLS

routing-table called-e164 RT_ISDN_TO_SIP_1
route T2 dest-interface IF_SIP_1 MAPPING_INCOMING_CALLS

routing-table called-e164 RT_SIP_TO_ISDN_0
route default dest-interface IF_ISDN_0 OUTBOUND

routing-table called-e164 RT_SIP_TO_ISDN_1
route default dest-interface IF_ISDN_1 OUTBOUND

mapping-table calling-pi to calling-e164 MAP_REMOVE_BLANK_CALLERID
map restricted to ""

mapping-table calling-e164 to calling-e164 MAP_LEADING_ZERO
map (3.%)-(.%)-(.%)-(.%)-(.%) to \1\2\3\4\5
map (3.%)-(.%)-(.%)-(.%) to \1\2\3\4
map (3.%)-(.%)-(.%) to \1\2\3
map (3.%)-(.%) to \1\2
map (3.%) to \1
map (.%)-(.%)-(.%)-(.%)-(.%) to 0\1\2\3\4\5
map (.%)-(.%)-(.%)-(.%) to 0\1\2\3\4
map (.%)-(.%)-(.%) to 0\1\2\3
map (.%)-(.%) to 0\1\2
map (.%) to 0\1

mapping-table calling-e164 to calling-name MAP_CID_TO_CNAME
map (.%) to \1

mapping-table calling-e164 to calling-e164 SET_CNPN
map 0(.%) to \1

mapping-table calling-type-of-number to calling-type-of-number TYPE
map default to national

mapping-table calling-numbering-plan to calling-numbering-plan PLAN
map default to isdn-telephony

complex-function MAPPING_INCOMING_CALLS
execute 1 MAP_REMOVE_BLANK_CALLERID
execute 2 MAP_LEADING_ZERO
execute 3 MAP_CID_TO_CNAME

complex-function OUTBOUND
execute 1 PLAN
execute 2 SET_CNPN
execute 3 TYPE

interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP_0
call-reroute emit
call-hold disable
diversion emit
caller-name
user-side-ringback-tone

interface isdn IF_ISDN_1
route call dest-table RT_ISDN_TO_SIP_1
call-reroute emit
call-hold disable
diversion emit
caller-name
user-side-ringback-tone

interface sip IF_SIP_0
bind context sip-gateway GW_SIP_0
route call dest-table RT_SIP_TO_ISDN_0
remote 192.168.1.77 5060
early-connect
early-disconnect
call-reroute accept
address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
trust remote

interface sip IF_SIP_1
bind context sip-gateway GW_SIP_1
route call dest-table RT_SIP_TO_ISDN_1
remote 192.168.1.77 5060
early-connect
early-disconnect
call-reroute accept
address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none
trust remote

context cs switch
no shutdown

authentication-service AS_ALL_LINES
realm 1 3CXPhoneSystem
username 10001 password dZ8edXkjFnM= encrypted
username 10002 password iAaKwKfxX4Q= encrypted

location-service LS_10001
domain 1 192.168.1.77

identity-group default

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10001

identity 10001

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES

registration outbound
registrar 192.168.1.77 5060
lifetime 300
register auto

location-service LS_10002
domain 1 192.168.1.77

identity-group default

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10002

identity 10002

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES

registration outbound
registrar 192.168.1.77 5060
lifetime 300
register auto

context sip-gateway GW_SIP_0

interface LAN
bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_0
bind location-service LS_10001
no shutdown

context sip-gateway GW_SIP_1

interface LAN
bind interface IF_IP_LAN context router port 5062

context sip-gateway GW_SIP_1
bind location-service LS_10002
no shutdown

port ethernet 0 0
medium 10 half
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_ISDN_0 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_ISDN_1 switch

port bri 0 1
no shutdown
 
Ciao,
che firmware è presente sul Patton? controlla di essere almeno alla release 6.6
 
Usi Telefoni Snom ? Su alcuni template se non utilizzi la versione di firmware supportata e certificata puoi incorrere in questo problema.
Aggiorna il firmware del Patton all'ultima release.
Controlla infine che il CallerID venga mostrato nella management console sotto "stato tronchi" quando hai la chiamata in ingresso. Questo è necessario per verificare che se il problema è nel Patton o nella configurazione dei telefoni.

Matteo Sala
Eurylink Shop Voip & Multimedia
3CX Premium Partner – 3CX Advanced Certified
http://www.eurylink.com
 

Inizia con 3CX – Admin

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