Patton SN4554 - No matching peer found

I’m successfully using a Patton SN4554/2BIS/EUI on my FreePBX but taking a look at /var/log/asterisk/full i can see the following message repeated frequently:

[2011-12-23 20:58:15] NOTICE[20787] chan_sip.c: Registration from ‘patton sip:[email protected]’ failed for ‘192.168.1.252:5060’ - No matching peer found

Resetting the patton I can’t receive any call form outside until I make a call form my PBX to myself.

Must be some error on Patton configuration…

Trunk Asterisk:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=192.168.1.252
username=101
secret=*************
insecure=port,invite
qualify=yes
type=peer
nat=no
port=5060
disallow=all
allow=ulaw

PATTON Config

Configurazione PATTON:

#----------------------------------------------------------------#

SN4554/2BIS/EUI

R5.7 2011-09-02 SIP

1970-01-23T10:00:14

SN/00A0BA04EF33

Generated configuration file

#----------------------------------------------------------------#

cli version 3.20
gui type basic
clock local default-offset +01:00
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary time.ien.it port 123 version 4

system

ic voice 0
low-bitrate-codec g729

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

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
no rate-limit

profile service-policy SP_WAN_IN
no rate-limit

profile napt NAPT_WAN

profile ppp default

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 call-progress-tone IT_Congestion
play 1 200 425 -12
pause 2 200
play 3 200 425 -12

profile tone-set default
map call-progress-tone congestion-tone IT_Congestion

profile tone-set IT
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 g711ulaw64k rx-length 20 tx-length 20
codec 2 g729 rx-length 20 tx-length 20
dtmf-relay signaling default

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router
rtp-port-range 16384 16481

interface IF_IP_WAN
ipaddress 192.168.1.252 255.255.255.0
use profile service-policy SP_WAN_IN in
use profile service-policy SP_WAN_OUT out
use profile napt NAPT_WAN
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_LAN
ipaddress unnumbered

subscriber ppp SUB_PPPOE
dial in
no multilink

context cs switch
digit-collection timeout 8
no digit-collection terminating-char
national-prefix 0
international-prefix 00

routing-table called-e164 SIP_TO_ISDN
route 1(.%) dest-interface IF_S0_00 TRUNC
route 2(.%) dest-interface IF_S0_01 TRUNC
route 0(.%) dest-service SER_HG_PSTN_FALLBACK TRUNC

mapping-table called-e164 to called-e164 TRUNC
map .(.%) to \1

interface isdn IF_S0_00
route call dest-interface IF_SIP
use profile tone-set IT

interface isdn IF_S0_01
route call dest-interface IF_SIP
use profile tone-set IT

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-table SIP_TO_ISDN
remote 192.168.1.250
early-disconnect
privacy
use profile tone-set IT

service hunt-group SER_HG_PSTN_FALLBACK
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_S0_00
route call 2 dest-interface IF_S0_01

context cs switch
no shutdown

authentication-service AUTH_SVC
username 101 password mFTXArREVtxnX+kX5xRjnw== encrypted

location-service LOCATION_SVC
imperative foreign
domain 1 192.168.1.250

identity patton
display-name patton

authentication outbound
  authenticate 1 authentication-service AUTH_SVC username 101

registration outbound
  registrar 192.168.1.250 5060
  lifetime 3600
  register auto
  retry-timeout on-system-error 10
  retry-timeout on-client-error 10
  retry-timeout on-server-error 10

context sip-gateway GW_SIP

interface eth0
bind interface IF_IP_WAN context router port 5060

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

port ethernet 0 0
bind interface IF_IP_WAN router

pppoe

session SES_PPPOE
  shutdown

port ethernet 0 0
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
uni-side user
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_S0_00 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
uni-side user
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_S0_01 switch

port bri 0 1
no shutdown

Anyone can help me please?

This is the only way I can solve the problem

It’s only a workaround but seems working good. I hope could be useful for someone else.

Merry Christmas!! :slight_smile:

Asterisk Trunk

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=192.168.1.252
insecure=port,invite
qualify=yes
type=friend
nat=no
port=5060
disallow=all
allow=ulaw&g729

Patton Config:

#----------------------------------------------------------------#

SN4554/2BIS/EUI

R5.7 2011-09-02 SIP

1970-01-24T07:17:57

SN/00A0BA04EF33

Generated configuration file

#----------------------------------------------------------------#

si autentica su Asterisk con IP evitando la registrazione

che non sono riuscito a far funzionare…

in uscita:

0 ==> Chiama la prima linea libera

1 ==> chiama ISDN 00

2 ==> chiama ISDN 01

cli version 3.20
gui type basic
clock local default-offset +01:00
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary time.ien.it port 123 version 4

system

ic voice 0
low-bitrate-codec g729

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

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
no rate-limit

profile service-policy SP_WAN_IN
no rate-limit

profile napt NAPT_WAN

profile ppp default

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 call-progress-tone IT_Congestion
play 1 200 425 -12
pause 2 200
play 3 200 425 -12

profile tone-set default
map call-progress-tone congestion-tone IT_Congestion

profile tone-set IT
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 g711ulaw64k rx-length 20 tx-length 20
codec 2 g729 rx-length 20 tx-length 20
dtmf-relay signaling default

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router
rtp-port-range 16384 16481

interface IF_IP_WAN
ipaddress 192.168.1.252 255.255.255.0
use profile service-policy SP_WAN_IN in
use profile service-policy SP_WAN_OUT out
use profile napt NAPT_WAN
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_LAN
ipaddress unnumbered

subscriber ppp SUB_PPPOE
dial in
no multilink

context cs switch
digit-collection timeout 8
no digit-collection terminating-char
national-prefix 0
international-prefix 00

routing-table called-e164 RT_TO_SIP
route .%T dest-interface IF_SIP_SERVICE

routing-table called-e164 SIP_TO_ISDN
route 1(.%) dest-interface IF_S0_00 TRUNC
route 2(.%) dest-interface IF_S0_01 TRUNC
route 0(.%) dest-service SER_HG_PSTN_FALLBACK TRUNC

mapping-table called-e164 to called-e164 TRUNC
map .(.%) to \1

interface isdn IF_S0_00
route call dest-table RT_TO_SIP
use profile tone-set IT

interface isdn IF_S0_01
route call dest-table RT_TO_SIP
use profile tone-set IT

interface sip IF_SIP_SERVICE
bind context sip-gateway GW_SIP
route call dest-table SIP_TO_ISDN
remote 192.168.1.250
early-connect
early-disconnect
use profile tone-set IT

service hunt-group SER_HG_PSTN_FALLBACK
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_S0_00
route call 2 dest-interface IF_S0_01

context cs switch
no shutdown

authentication-service AUTH_SVC
username 101 password mFTXArREVtxnX+kX5xRjnw== encrypted

location-service LOCATION_SVC
domain 1 192.168.1.250

identity 101
display-name 101

context sip-gateway GW_SIP

interface IF_SIP
bind interface IF_IP_WAN context router port 5060

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

port ethernet 0 0
bind interface IF_IP_WAN router

pppoe

session SES_PPPOE
  shutdown

port ethernet 0 0
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
uni-side user
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_S0_00 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
uni-side user
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_S0_01 switch

port bri 0 1
no shutdown