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....
Version imprimable
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....
Bonjour, il faudrait que tu fournisses la cli pendant la tentative d'appel sortant
== 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'
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-...-son-trunk-sip
http://adminrzo.blogspot.fr/2012/03/...n4554-r52.html
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
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
Affiche aussi le sip show registry et le sip show peers
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]
#----------------------------------------------------------------#
# #
# 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=$(cli.major>=4|Trinity|Sm artWare);subDirTrinity=$(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|Sm artWare);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
problème résolu; c'était au niveau du serveur Asterisk...