Patton SN4120 trunk not register on freepbx

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Patton SN4120 trunk not register on freepbx

Postby asyscom » Mon Dec 14, 2015 9:30 am

Hi, i'm try to configure an patton sn4120/2bri but on freepbx the trunks remain unregistered.

In freepbx dashboard the 2 trunk are displayed as offline and, also in consolle if i digit "sip show peers" the trunk are unregister while all other extensions are registered

I've tried to stay in asterisk consolle and restart patton to see any error but nothing, seems that patton don't make registration request on freepbx.

In follow my conf and trunks
Ip PBX 192.168.16.14
Ip Patton 192.168.16.15

#----------------------------------------------------------------
#
# http://www.patton-smartnode-configuration.com
# Asterweb Srl - Milan - Italy
# info@asterweb.org
#
# Firmware 6.x
# Generated configuration file 2015-12-14 14:16:22
#
#----------------------------------------------------------------

cli version 3.20
clock local default-offset +01:00
dns-client server 192.168.16.10
dns-client server 8.8.4.4
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.16.14 port 123 version 4
system hostname SN4120/2bis

system
ic voice 0

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

profile ppp default

profile call-progress-tone IT_Dialtone
play 1 200 425 -12
pause 2 200
play 3 600 425 -12
pause 4 1000
play 5 200 425 -12
pause 6 200
play 7 600 425 -12
pause 8 1000
play 9 200 425 -12
pause 10 200

profile call-progress-tone IT_Alertingtone
play 1 1000 425 -12
pause 2 4000
play 3 1000 425 -12
pause 4 4000
play 5 1000 425 -12
pause 6 4000

profile call-progress-tone IT_Busytone
play 1 500 425 -12
pause 2 500
play 3 500 425 -12
pause 4 500
play 5 500 425 -12
pause 6 500

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
codec 3 g729 rx-length 20 tx-length 20
fax transmission 1 relay t38-udp

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router
interface LAN
ipaddress 192.168.16.15 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 0.0.0.0 0.0.0.0 192.168.16.1 0

context cs switch
digit-collection timeout 3
no digit-collection terminating-char
national-prefix 0
international-prefix 00

routing-table called-e164 INBOUND_0
route .%T3 dest-interface IF_SIP_SERVICE_0

routing-table called-e164 INBOUND_1
route .%T3 dest-interface IF_SIP_SERVICE_1

routing-table called-e164 OUTBOUND_0
route default dest-service BOUND_PSTN_0

routing-table called-e164 OUTBOUND_1
route default dest-service BOUND_PSTN_1

interface isdn IF_PSTN_0
route call dest-table INBOUND_0
use profile tone-set IT

interface isdn IF_PSTN_1
route call dest-table INBOUND_1
use profile tone-set IT

interface sip IF_SIP_SERVICE_0
bind context sip-gateway GW_SIP_0
route call dest-table OUTBOUND_0
remote 192.168.16.14
early-disconnect
privacy

interface sip IF_SIP_SERVICE_1
bind context sip-gateway GW_SIP_1
route call dest-table OUTBOUND_1
remote 192.168.16.14
early-disconnect
privacy

service hunt-group BOUND_PSTN_0
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
drop-cause user-busy
route call 1 dest-interface IF_PSTN_0

service hunt-group BOUND_PSTN_1
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
drop-cause user-busy
route call 1 dest-interface IF_PSTN_1

context cs switch
no shutdown

authentication-service AUTH_SVC
username patton password 1234
username patton1 password 1234

location-service LOCATION_SVC_0
domain 1 192.168.16.14

identity patton
authentication outbound
authenticate 1 authentication-service AUTH_SVC username patton

registration outbound
registrar 192.168.16.14
lifetime 3600
register auto

location-service LOCATION_SVC_1
domain 1 192.168.16.14

identity patton1
authentication outbound
authenticate 1 authentication-service AUTH_SVC username patton1

registration outbound
registrar 192.168.16.14
lifetime 3600
register auto


context sip-gateway GW_SIP_0
interface IF_GW_SIP_0
bind interface LAN context router port 5060

context sip-gateway GW_SIP_0
bind location-service LOCATION_SVC_0
no shutdown

context sip-gateway GW_SIP_1
interface IF_GW_SIP_1
bind interface LAN context router port 5061

context sip-gateway GW_SIP_1
bind location-service LOCATION_SVC_1
no shutdown

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

port bri 0 0
clock auto
encapsulation q921

q921
permanent-layer2
uni-side auto
protocol pp
encapsulation q931

q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_PSTN_0 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
permanent-layer2
uni-side auto
protocol pp
encapsulation q931

q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_PSTN_1 switch

port bri 0 1
no shutdown


E questi i 2 trunk

username=patton
secret=1234
type=friend
host=dynamic
port=5060
nat=no
directmedia=no
defaultip=192.168.16.15
context=from-pstn
insecure=port,invite
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=alaw&ulaw


username=patton1
secret=1234
type=friend
host=dynamic
port=5061
nat=no
directmedia=no
defaultip=192.168.16.15
context=from-pstn
insecure=port,invite
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=alaw&ulaw


Thanks in advance Davide
asyscom
Newsterisk
 
Posts: 21
Joined: Mon Mar 23, 2009 3:18 am

Re: Patton SN4120 trunk not register on freepbx

Postby asyscom » Wed Dec 16, 2015 2:13 am

Update:

After a lot of attempts i've doscovered that if I create a pjsip trunk the patton register on freepbx without problem while with normal sip trunk i have this error message:

[2015-12-15 17:12:07] NOTICE[1961]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '<sip:PattonUser2@192.168.16.14>' failed for '192.168.16.15:5061' (callid: 9687bc8ce242b483) - No matching endpoint found

Why says about pjsip if i've create a sip trunk?

Pjsip trunk wiht patton can work well?

Thanks in advance
Davide
asyscom
Newsterisk
 
Posts: 21
Joined: Mon Mar 23, 2009 3:18 am


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