Problema chiamate in uscita freepbx e pattn sn4171

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asyscom
Utente
Messaggi: 33
Iscritto il: mercoledì 24 febbraio 2010, 13:22

Problema chiamate in uscita freepbx e pattn sn4171

Messaggio da asyscom » mercoledì 10 gennaio 2018, 10:12

Ciao a tutti,
ho un problema che mi sa bloccando il progetto di messa in operatività di un nuovo centralino per una azienda.
La mia configurazione è freepbx FreePBX 14.0.1.20 + patton SN4171

La situazione è semplice, in ingresso funziona tutto regolarmente sia chiamate dirette tramite selzione passante che IVR, il problema sono le chiamate in uscita, da dialplan le istruzioni sono instradate correttamente solo che l'errore è il seguente, il classico 503

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/gatewaypri1/01875295100
-- Got SIP response 503 "Service Unavailable" back from 192.168.10.132:5060
-- SIP/gatewaypri1-0000009b is circuit-busy

Di seguito le configurazioni:

Patton:

#----------------------------------------------------------------
# Patton Electronics Company
# Wizard generated config file
# Name:
# Trinity SN4170 / SN4970 / SN4971 Basic Setup
# Description:
# This sets up your Trinity SN4170 or SN4970/71 with either an IPPBX or an ITSP SIP Trunk.
#----------------------------------------------------------------
cli version 4.0

clock local default-offset +01:00

profile aaa DEFAULT
method 1 local rule required
method 2 none rule required

console
use profile aaa DEFAULT

telnet-server
use profile aaa DEFAULT
no shutdown

ssh-server
use profile aaa DEFAULT
no shutdown

snmp-server
shutdown

web-server http
use profile aaa DEFAULT
no shutdown

system
clock-source 1 e1t1 0 0

ntp
server 0.patton.pool.ntp.org
server 1.patton.pool.ntp.org
server 2.patton.pool.ntp.org
server 3.patton.pool.ntp.org
no shutdown

dns-client
name-server 192.168.10.10

profile tls DEFAULT
no authentication incoming
no authentication outgoing
private-key pki:private-key/DEFAULT
own-certificate 1 pki:own-certificate/DEFAULT


profile call-progress-tone IT_Dialtone
play 200 425 -12
pause 200
play 600 425 -12
pause 1000


profile call-progress-tone IT_Alertingtone
play 1000 425 -12
pause 4000


profile call-progress-tone IT_Busytone
play 500 425 -12
pause 500

profile tone-set default

profile tone-set IT
map call-progress-tone dial-tone IT_Dialtone
map call-progress-tone ringback-tone IT_Alertingtone
map call-progress-tone busy-tone IT_Busytone
map call-progress-tone release-tone IT_Busytone
map call-progress-tone congestion-tone IT_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 20 tx-length 20
fax transmission 3 bypass g711ulaw64k rx-length 20 tx-length 20
fax bypass-method signaling
modem transmission 1 bypass g711alaw64k rx-length 20 tx-length 20
modem transmission 2 bypass g711ulaw64k rx-length 20 tx-length 20
modem bypass-method signaling



profile pstn DEFAULT

profile sip DEFAULT

context ip ROUTER

interface WAN
ipaddress WAN 192.168.10.132 255.255.255.0

routing-table DEFAULT
route 0.0.0.0/0 gateway 192.168.10.85 metric 0

profile ppp DEFAULT

context bridge

context cs SWITCH
no shutdown

routing-table called-e164 RT_ISDN_TO_SIP
route T2 dest-interface IF_SIP

interface isdn IF_ISDN_00
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
user-side-ringback-tone
caller-name

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-interface IF_ISDN_00
remote 192.168.10.130
trust remote

authentication-service AUTH_SRV
username gatewaypri1 password tEtEnewairlTY7j/OmBBeg== encrypted

location-service SER_LOC
domain 1 192.168.10.130
match-any-domain

identity-group DEFAULT

authentication inbound
authenticate 1 authentication-service AUTH_SRV username gatewaypri1

registration inbound

identity gatewaypri1 inherits DEFAULT

context sip-gateway GW_SIP
bind location-service SER_LOC

interface SIP
transport-protocol udp+tcp 5060
no transport-protocol tls

bind ipaddress ROUTER WAN WAN

context sip-gateway GW_SIP
no shutdown

port ethernet 0 0
bind interface ROUTER WAN
no shutdown

port e1t1 0 0
port-type e1
clock auto
framing crc
encapsulation q921

q921
permanent-layer2
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_00

port e1t1 0 0
no shutdown


Sip User:
[gatewaypri1]
disallow=all
context=from-trunk
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=gatewaypri1
secret=gatewaypri1
host=192.168.10.132
nat=force_rport,comedia
allow=alaw
allow=ulaw

Trunksip:
context=from-trunk
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=gatewaypri1
secret=gatewaypri1
host=192.168.10.132
nat=yes
disallow=all
allow=alaw&ulaw


In ultima analisi ho fatto il debug sul patton ma non vedo nessuna chiamata in uscita pertanto l'errore non viene dato da Telecom ma proprio da patton.

Grazie in aticicpo

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