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Voir la version complète : Probleme configuration Patton SN4554



rock17
24/11/2010, 08h12
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.

tomarch
24/11/2010, 09h16
fait un essai en modifiant tes trunks avec ces paramètres :

type=peer
context=from-trunk

rock17
24/11/2010, 12h22
c'est exactement la même chose

Merci

== 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/5062-000000c4", "Received incoming SIP connection from unknown peer to 10001") in new stack
-- Executing [10001@from-sip-external:2] Set("SIP/5062-000000c4", "DID=10001") in new stack
-- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000c4", "0?from-trunk,10001,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5062-000000c4", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2010-11-24 14:23:12.000 RET.
-- Executing [s@from-sip-external:3] Answer("SIP/5062-000000c4", "") in new stack
-- Executing [s@from-sip-external:4] Wait("SIP/5062-000000c4", "2") in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/5062-000000c4", "ss-noservice") in new stack
-- <SIP/5062-000000c4> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/5062-000000c4", "congestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/5062-000000c4", "5") in new stack
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/5062-000000c4'
-- Executing [h@from-sip-external:1] NoOp("SIP/5062-000000c4", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/5062-000000c4", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5062-000000c4", "0?from-trunk,s,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5062-000000c4", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2010-11-24 14:23:23.000 RET.
-- Executing [s@from-sip-external:3] Answer("SIP/5062-000000c4", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/5062-000000c4'

rock17
24/11/2010, 12h24
Actuellement je suis en R5.6 sur la Patton SN 4554, j'ai vu des tuto avec des version R4.2 je vais downgrader en R4.2 je fait des essai et je reviens.

et


je n'arrive pas a la downgrader

:gratgrat:

jean
24/11/2010, 13h22
je trouve ces deux lignes bizarre:


-- Executing [10001@from-sip-external:3] Goto("SIP/5062-000000c4", "s,1") in new stack
-- Goto (from-sip-external,s,1)

y'a pas de raisons de faire un goto "s" - c'est ca qui déclenche le message à la con et le raccroché

jean
24/11/2010, 14h04
et si tu regardes plus en détail,

l'appel arrive sur l'ext 1001, bascule via goto sur s (normalement utilisé quand il n'y a pas de no d'extension fourni), tombe sur un congestion, bascule sur h (logique, raccroché), et refait un goto "s".... c'est pas sain, il n'y a pas de logique à ce que le h renvoie sur le s (ni qu'une exten valide renvoie sur le s)

rock17
26/11/2010, 12h17
finalement ca fonctionne, tomarch avait presque la reponse, c'etait tout simplement un probleme de configuration du Trunk, j'en ai profité pour faire un tuto:

http://www.asterisk-france.org/showthread.php/372-Tuto-configuration-Patton-SN4554-R5.2-sur-Asterisk-%28Trixbox%29