mes appels ne sortent pas

[dulcinee14]

Bonjour,
j''ai configuré mon asterisk avec la patton 4660. je reçois les appels, mais je ne peux pas en émettre..please help me, je dois terminer ce projet....

[koma87]

Bonjour, il faudrait que tu fournisses la cli pendant la tentative d'appel sortant

[dulcinee14]

== Using SIP RTP CoS mark 5
-- Executing [708535334@SocieteX:1] Answer("SIP/101-00000004", "") in new stack
-- Executing [708535334@SocieteX:2] Set("SIP/101-00000004", "CHANNEL(LANGUAGE)=fr") in new stack
-- Executing [708535334@SocieteX:3] Dial("SIP/101-00000004", "SIP/TELOGIK/708535334,15") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/TELOGIK/708535334
-- Got SIP response 400 "Bad Request" back from 10.10.1.198:5060
-- SIP/TELOGIK-00000005 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [708535334@SocieteX:4] Hangup("SIP/101-00000004", "") in new stack
== Spawn extension (SocieteX, 708535334, 4) exited non-zero on 'SIP/101-00000004'

[koma87]

Il faudrait aussi que tu donnes tes fichiers de config de la patton et de ton sip.conf

Je suppose que la config est la même que sur tous les posts que tu as fait précédemment ?

Tu as bien mis un champ username dans le SIP.conf ?

Regarde sur les 2 liens suivant, les exemples sont bien parlant.

http://www.senetel.fr/actualites/85-asterisk-episode-2-parametrer-son-trunk-sip

http://adminrzo.blogspot.fr/2012/03/tuto-configuration-patton-sn4554-r52.html

[dulcinee14]

Mon fichier sip.conf

[2000]
type=friend
secret=passer
port=5060
nat=yes
host=dynamic
context=SocieteX
dial=SIP/2000
dtmfmode=auto
insecure=port,invite
context=SocieteX
quality=yes
canreinvite=no
disallow=all
;allow=g729
allow=ulaw
allow=alaw
;allow=gsm
language=fr


[Patton]
type=friend
host=10.10.x.x
context=SocieteX
fromdomain=10.10.x.x
;nat=yes
;username=2000
;secret=passer
;insecure=invite,port
permit=10.10.x.x/255.255.255.0
;quality=yes
;disallow=all
;allow=g729
;allow=ulaw
;allow=alaw
;allow=gsm
;canreinvite=no

mon extensions.conf
exten => _88xxxxxxx,1,Answer()
exten => _88xxxxxxx,2,Set(CHANNEL(LANGUAGE)=fr)
exten => _88xxxxxxx,3,Dial(SIP/Patton/${EXTEN}:1,15)
exten => _88xxxxxxx,4,Hangup()

je me suis inspiré des fichiers effectuvement. Je reçois les appels, juiste que je n'arrve pas à émettre.
merci

[jean]

il faut mettre les traces sur la patton:
telnet ipdelapatton

login,

puiis

enable
configure
debug context sip-gateway transport detail 1
debug context sip-gateway signaling detail 1
debug ccisdn signaling
debug context sip-gateway error detail 3

si pas assez clair, monter les niveaux de debug des 2 premieres lignes

[koma87]

Affiche aussi le sip show registry et le sip show peers

[dulcinee14]

sip show registry
Host dnsmgr Username Refresh State Reg.Time
10.10.1.198:5060 N 2000 105 Registered Tue, 03 Mar 2015 14:32:45
1 SIP registrations.

sip show peers
Name/username Host Dyn Forcerport ACL Port Status
101/101 10.10.1.146 D N 52853 Unmonitored
102/102 10.10.1.146 D N 52853 Unmonitored
2000/2000 10.10.1.198 D N 5060 Unmonitored
210/210 (Unspecified) D N 0 Unmonitored
211/211 (Unspecified) D N 0 Unmonitored
patton 10.10.1.198 N A 5060 Unmonitored
trunk_vers_asterisk_gui (Unspecified) D N 0 Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 3 offline]

[dulcinee14]

#----------------------------------------------------------------#
# #
# SN4661/2BIS2JS2JO8V/EUI #
# R6.3 2013-05-01 H323 RBS SIP #
# 2015-03-03T14:54:01 #
# SN/00A0BA0A9E3C #
# Generated configuration file #
# #
#----------------------------------------------------------------#

cli version 3.20
administrator stage password FL1ch2SI5ohKCcDtA1yDCw== encrypted
clock local default-offset +00:00
timer PROVISIONING now + 3 minutes "provisioning execute PF_PROVISIONING_CONFIG"
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary pool.ntp.org port 123 version 4
system hostname PATTON

system

ic voice 0

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

profile napt NAPT_WAN

profile ppp default

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

profile voip voip1
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default
profile pstn PSTN01

profile ringing-cadence default
play 1 1000
pause 2 4000

profile sip default
autonomous-transitioning

profile sip px01
autonomous-transitioning

profile aaa default
method 1 local
method 2 none

profile provisioning PF_PROVISIONING_CONFIG
destination configuration
location 1 http://redirect.patton.com/$(system.mac);mac=$(system.mac);serial=$(system.se rial);hwMajor=$(system.hw.major);hwMinor=$(system. hw.minor);swMajor=$(system.sw.major);swMinor=$(sys tem.sw.minor);swDate=$(system.sw.date);productName =$(system.product.name);cliMajor=$(cli.major);cliM inor=$(cli.minor);osName=$(cli.major>=4|Trinity|SmartWare);subDirTrinity=$(cli.major>=4|/Trinity);subDirSmartWare=$(cli.major<4|/SmartWare);dhcp66=$(dhcp.66);dhcp67=$(dhcp.67)
location 2 $(dhcp.66)
location 3 $(dhcp.66)/$(system.mac).cfg
location 4 http://$(dhcp.66)/$(dhcp.67)
location 5 http://$(dhcp.66)/$(system.mac).cfg
location 6 tftp://$(dhcp.66)/$(dhcp.67)
location 7 tftp://$(dhcp.66)/$(system.mac).cfg
location 8 http://redirect.patton.com/$(system.mac);mac=$(system.mac);serial=$(system.se rial);hwMajor=$(system.hw.major);hwMinor=$(system. hw.minor);swMajor=$(system.sw.major);swMinor=$(sys tem.sw.minor);swDate=$(system.sw.date);productName =$(system.product.name);cliMajor=$(cli.major);cliM inor=$(cli.minor);osName=$(cli.major>=4|Trinity|SmartWare);subDirTrinity=$(cli.major>=4|/Trinity);subDirSmartWare=$(cli.major<4|/SmartWare);dhcp66=$(dhcp.66);dhcp67=$(dhcp.67)
location 9 $(dhcp.66)
location 10 $(dhcp.66)/$(system.mac).cfg
location 11 http://$(dhcp.66)/$(dhcp.67)
location 12 http://$(dhcp.66)/$(system.mac).cfg
location 13 tftp://$(dhcp.66)/$(dhcp.67)
location 14 tftp://$(dhcp.66)/$(system.mac).cfg
activation reload immediate

context ip router

interface WAN
ipaddress 10.10.1.198 255.255.255.0
icmp router-discovery
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

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

context cs switch

routing-table called-e164 RT_OUT
route .T dest-interface IF_SIP
route .%T dest-interface IF_AN1
route 00[1-9].T dest-interface IF_AN1
route 7[0678]....... dest-interface IF_AN1
route 3388..... dest-interface IF_AN1
route 3[03]....... dest-interface IF_AN1
route 338...... dest-interface IF_SIP

routing-table called-e164 RT_IN
route .%T dest-interface IF_SIP
route 1[0123456789] dest-interface IF_SIP

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-interface IF_AN1
remote 10.10.1.146 5060
no early-proceeding
no call-transfer pull-in
address-translation outgoing-call to-header user-part call host-part fix 10.10.1.146 5060

interface fxs IF_TEL
route call dest-interface IF_SIP

interface fxs IF_TEL2
route call dest-interface IF_SIP

interface fxo IF_AN1
route call dest-interface IF_SIP
no disconnect-signal loop-break
ring-number on-caller-id
dial-after timeout 5

interface fxo IF_AN2
disconnect-signal loop-break

service hunt-group SERVICE1
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_AN1

context cs switch
no shutdown

authentication-service SERV_ASTERISK_AUTH
username 2000 password FL1ch2SI5ohKCcDtA1yDCw== encrypted

location-service REG_SIP
domain 1 10.10.1.146 5060
match-any-domain

identity 2000

authentication outbound
authenticate 1 authentication-service SERV_ASTERISK_AUTH username 2000

registration outbound
registrar 10.10.1.146
lifetime 3600
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10
nat-traversal minimal

registration inbound
lifetime default 3600 min 1 max 4294967295

call outbound
use profile tone-set default
use profile voip default
use profile sip default

call inbound
use profile tone-set default
use profile voip default
use profile sip default

context sip-gateway GW_SIP

interface IF_IP_GW
bind interface WAN context router port 5060

context sip-gateway GW_SIP
bind location-service REG_SIP
no shutdown

sip
rport

port ethernet 0 0
encapsulation ip
bind interface WAN router
no shutdown

port ethernet 0 1
encapsulation ip
bind interface LAN router
no shutdown

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

port fxs 0 1
encapsulation cc-fxs
bind interface IF_TEL2 switch
no shutdown

port fxo 0 0
encapsulation cc-fxo
bind interface IF_AN1 switch
no shutdown

port fxo 0 1
encapsulation cc-fxo
bind interface IF_AN2 switch
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

port bri 0 0
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

port bri 0 1
shutdown

[dulcinee14]

problème résolu; c'était au niveau du serveur Asterisk...