Probleme configuration Patton SN4554
Bonjour a tous.
Nous avons une passerelle Patton SN4554 (192.168.12.164) ou arrive 2 T0 (avec 10 SDA)
Puis un IPBX Asterisk (Trixbox) (192.168.12.163).
Le problème est qu'on arrive bien a émettre des appels mais pas a en recevoir.
Voici la conf du Patton (prise sur un forum)
cli version 3.20
clock local offset -04:00
dns-client server 192.168.23.5
dns-relay
webserver port 80 language fr
sntp-client
sntp-client server primary 192.168.15.254 port 123 version 4
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 1000 440 0
profile call-progress-tone defaultAlertingtone
play 1 1500 440 -7
pause 2 3500
profile call-progress-tone defaultBusytone
play 1 500 440 -7
pause 2 500
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
fax transmission 1 relay t38-udp
profile pstn default
profile sip default
profile aaa default
method 1 local
method 2 none
context ip router
interface IF_IP_LAN
ipaddress 192.168.12.164 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
route 0.0.0.0 0.0.0.0 192.168.12.254 0
context cs switch
national-prefix 0
international-prefix 00
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
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 (.%) to \1
complex-function MAPPING_INCOMING_CALLS
execute 1 MAP_REMOVE_BLANK_CALLERID
execute 2 MAP_LEADING_ZERO
interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP_0
caller-name
user-side-ringback-tone
interface isdn IF_ISDN_1
route call dest-table RT_ISDN_TO_SIP_1
caller-name
user-side-ringback-tone
interface sip IF_SIP_0
bind context sip-gateway GW_SIP_0
route call dest-interface IF_ISDN_0
remote 192.168.12.163
early-disconnect
address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
address-translation outgoing-call diversion-header host-part call
address-translation incoming-call calling-e164 fix 0596646867
interface sip IF_SIP_1
bind context sip-gateway GW_SIP_1
route call dest-interface IF_ISDN_1
remote 192.168.12.163
early-disconnect
address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
address-translation incoming-call calling-e164 fix 0596646867
context cs switch
no shutdown
authentication-service AS_ALL_LINES
realm 1 ELASTIX
username 10000 password LbFCDNv4/Fk= encrypted
username 10001 password dZ8edXkjFnM= encrypted
location-service LS_10000
domain 1 192.168.12.163
domain 2 192.168.23.25
identity-group default
authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10000
identity 10000
authentication outbound
authenticate 1 authentication-service AS_ALL_LINES
registration outbound
registrar 192.168.12.163
lifetime 300
register auto
location-service LS_10001
domain 1 192.168.12.163
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 192.168.12.163
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_10000
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_10001
no shutdown
port ethernet 0 0
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
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
Puis sur Asterisk on a créé 2 trunk
nom du trunk: 10000
Detail du peer:
canreinvite=no
context=from-pstn
host=192.168.12.164
dtmfmode=auto
port=5060
qualify=yes
type=friend
username=10000
secret=10000
allow=alaw&ulaw&gsm
nom du trunk: 10001
Detail du peer:
canreinvite=no
context=from-pstn
host=192.168.12.164
dtmfmode=auto
port=5060
qualify=yes
type=friend
username=10001
secret=10001
allow=alaw&ulaw&gsm
on a creer une route entrant qui route tout vers le poste 6701
et une route sortant generale vers la patton
lorsqu'on appelle un SDA depuis l'exterieur on arrive bien a l'IPBX mais pas sur le poste 6701 on arrive sur un message comme quoi le poste n'existe pas.
Voici les log asterisk: (asterisk -r)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [10001@from-sip-external:1] NoOp("SIP/5062-000000b7", "Received incoming SIP connection from unknown peer to 10001") in new stack
-- Executing [10001@from-sip-external:2] Set("SIP/5062-000000b7", "DID=10001") in new stack
-- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000b7", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000b7", "0?from-trunk,10001,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5062-000000b7", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2010-11-24 09:53:42.000 RET.
-- Executing [s@from-sip-external:3] Answer("SIP/5062-000000b7", "") in new stack
-- Executing [s@from-sip-external:4] Wait("SIP/5062-000000b7", "2") in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/5062-000000b7", "ss-noservice") in new stack
-- <SIP/5062-000000b7> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/5062-000000b7", "congestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/5062-000000b7", "5") in new stack
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/5062-000000b7'
-- Executing [h@from-sip-external:1] NoOp("SIP/5062-000000b7", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/5062-000000b7", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/5062-000000b7", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000b7", "0?from-trunk,s,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5062-000000b7", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2010-11-24 09:53:51.000 RET.
-- Executing [s@from-sip-external:3] Answer("SIP/5062-000000b7", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/5062-000000b7'
trixbox1*CLI>
j'avoue que c'est un peu du charabia pour moi
merci de votre aide.
