configuration des ports Patton FXS avec Asterisk

[amine2018]

Bonjour
Est ce que quelqu'un pourrai m'aider a configurer les port FXS de mon patton afin de fonctionner avec Asterisk.
J'ai enregistrer le SIP d'un FXS mais je trouve pas comment m'en servir pour passer des appels interne.

voici ma config patton :
#----------------------------------------------------------------#
# #
# SN4114/2JS2JO/EUI #
# R6.4 2013-07-12 H323 SIP FXS FXO #
# # Generated configuration file #
# #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +00:00
dns-client server 192.168.1.245
webserver port 80 language en
sntp-client server primary 129.132.2.21 port 123 version 4

system

ic voice 0
low-bitrate-codec g729

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 call-progress-tone FR_Dialtone
play 1 5000 440 -10

profile call-progress-tone FR_Alertingtone
play 1 1500 440 -10
pause 2 3500

profile call-progress-tone FR_Busytone
play 1 500 440 -10
pause 2 500

profile tone-set default
profile tone-set TS_FRANCE
map call-progress-tone dial-tone FR_Dialtone
map call-progress-tone ringback-tone FR_Alertingtone
map call-progress-tone busy-tone FR_Busytone
map call-progress-tone release-tone FR_Busytone
map call-progress-tone congestion-tone FR_Busytone

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
fax transmission 2 bypass g711alaw64k rx-length 10 tx-length 10

profile pstn default

profile ringing-cadence default
play 1 1000
pause 2 4000

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_LAN
ipaddress 192.168.1.14 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_WAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 0.0.0.0 0.0.0.0 192.168.1.245 0

context cs switch
digit-collection timeout 2

routing-table called-e164 RT_DIGITCOLLECTION
route .T dest-interface IF_SIP_PBX MT_EXT_TO_NAME

routing-table called-e164 RT_TO_FXS
route 201 dest-interface IF_FXS_1
route 202 dest-interface IF_FXS_2

mapping-table calling-e164 to calling-name MT_EXT_TO_NAME
map 201 to "Fname user 1 Lname user 1"
map 202 to "Fname user 2 Lname user 2"

interface sip IF_SIP_1
bind context sip-gateway GW_SIP_1
route call dest-interface IF_FXO_1
remote 192.168.1.250 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none

interface sip IF_SIP_2
bind context sip-gateway GW_SIP_2
route call dest-interface IF_FXO_2
remote 192.168.1.250 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10003 host-part to-header target-param none

interface sip IF_SIP_PBX
bind context sip-gateway GW_SIP_ALL_EXTENSIONS
route call dest-table RT_TO_FXS
remote 192.168.1.250 5060
local 192.168.1.14

interface fxs IF_FXS_1
route call dest-table RT_DIGITCOLLECTION
message-waiting-indication stutter-dial-tone
message-waiting-indication frequency-shift-keying
call-transfer
caller-id-presentation pre-ring
subscriber-number 201

interface fxs IF_FXS_2
route call dest-table RT_DIGITCOLLECTION
message-waiting-indication stutter-dial-tone
message-waiting-indication frequency-shift-keying
call-transfer
caller-id-presentation pre-ring
subscriber-number 202

interface fxo IF_FXO_1
route call dest-interface IF_SIP_1
loop-break-duration min 200 max 1000
disconnect-signal loop-break
disconnect-signal busy-tone
dial-after timeout 1
use profile tone-set TS_FRANCE

interface fxo IF_FXO_2
route call dest-interface IF_SIP_2
loop-break-duration min 200 max 1000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing

context cs switch
no shutdown

authentication-service AS_ALL_EXTENSIONS
username 201 password e1Q8XrEKJBg= encrypted
username 202 password i737zmKcO9A= encrypted

authentication-service AS_ALL_LINES
username 10002 password iAaKwKfxX4Q= encrypted
username 10003 password ySyWxUvSc/c= encrypted

location-service LS_ALL_EXTENSIONS
domain 1 192.168.1.250 5060

identity-group default
identity 201

authentication outbound
authenticate 1 authentication-service AS_ALL_EXTENSIONS username 201

registration outbound
registrar 192.168.1.250 5060
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

identity 202

authentication outbound
authenticate 1 authentication-service AS_ALL_EXTENSIONS username 202

registration outbound
registrar 192.168.1.250 5060
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LS_1
domain 1 192.168.1.250 5060

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 username 10002

registration outbound
registrar 192.168.1.250 5060
lifetime 300
register auto

location-service LS_2
domain 1 192.168.1.250 5060

identity-group default

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10003

identity 10003

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10003

registration outbound
registrar 192.168.1.250 5060
lifetime 300
register auto

context sip-gateway GW_SIP_1

interface LAN
bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_1
bind location-service LS_1
no shutdown

context sip-gateway GW_SIP_2

interface LAN
bind interface IF_IP_LAN context router port 5062

context sip-gateway GW_SIP_2
bind location-service LS_2
no shutdown

context sip-gateway GW_SIP_ALL_EXTENSIONS

interface LAN

context sip-gateway GW_SIP_ALL_EXTENSIONS
bind location-service LS_ALL_EXTENSIONS
no shutdown

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

port fxs 0 0
encapsulation cc-fxs
bind interface IF_FXS_1 switch
no shutdown

port fxs 0 1
encapsulation cc-fxs
bind interface IF_FXS_2 switch
no shutdown

port fxo 0 0
encapsulation cc-fxo
bind interface IF_FXO_1 switch
no shutdown

port fxo 0 1
encapsulation cc-fxo
bind interface IF_FXO_2 switch
no shutdown


Merci

[jean]

Ah patton, c'est pas du matos de touriste ;-)

Qu'est ce qui ne va pas exactement ? peux tu décrire ce qui se passe quand tu essaies de faire un appel, qu'est ce qui apparait sur la console asterisk ?

[amine2018]

Ah patton, c'est pas du matos de touriste ;-)

Qu'est ce qui ne va pas exactement ? peux tu décrire ce qui se passe quand tu essaies de faire un appel, qu'est ce qui apparait sur la console asterisk ?

Justement, ça se trouve que je suis un touriste dans ce domaine .....
En fait, je voudrai savoir comment la configuration des port fxs sur asterisk ... je,voudrai passer des appel interne (sip) avec d'autre poste mais je ne sais pas comment faire marcher ça. j'ai créer un trunk pour le fxs (201) qui s'enregistre sur asterisk, qu'est ce qu'il faure faire après ....

[jean]

dans ton sip.conf, tu crées des extensions comme pour un poste sip normal. tu y indiques un contexte. dans ce contexte, tu vas recevoir les appels entrants, tu as des trucs genre exten => s,1,... pour les appels sans caller id, et exten => _0.,1,.... pour les autres (si tu as la pres du numero)

pour les appels sortants tu fais un dial (SIP/extensiondelapasserelle/numeroaappeler)

pour plus d'infos, il faut être plus précis !