Hello,
I have the patton in object configured (I don’t think correctly of these points), from freepbx I can call without problems but when I try to receive from the asterisk cli I don’t see anything.
This is the patton running conf, do you have any idea where the problem is?
#----------------------------------------------------------------
# Patton Electronics Company
# Wizard generated config file
# Name:
# SN5530 / SN4130 Trinity ISDN BRI Basic Setup
# Description:
# This sets up your Trinity SN5530/SN4130 with either an IPPBX or an ITSP SIP Trunk.
#----------------------------------------------------------------
cli version 4.0
no clock local default-offset
rtp-port-range 6000 9999
profile aaa DEFAULT
method 1 local
method 2 none
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
web-server https
use profile aaa DEFAULT
no shutdown
system
clock-source 1 bri 0 0
clock-source 2 bri 0 1
clock-source 3 bri 0 2
clock-source 4 bri 0 3
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
profile napt NAPT_WAN
profile dhcp-server DHCPS_LAN
network 192.168.1.0/24
lease 24 hours
default-router 192.168.1.1
domain-name-server 192.168.1.1
include 192.168.1.10 192.168.1.99
profile tls DEFAULT
authentication incoming
authentication outgoing
private-key pki:private-key/DEFAULT
own-certificate 1 pki:own-certificate/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_Busyton
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 DHCP
profile ppp DEFAULT
context bridge
context cs SWITCH
no shutdown
routing-table called-e164 RT_ISDN_TO_SIP
route .T dest-service SER_SIP_SER
interface isdn IF_ISDN_00
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
interface isdn IF_ISDN_01
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
interface isdn IF_ISDN_02
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
interface isdn IF_ISDN_03
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
service hunt-group SRV_HG
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_ISDN_00
route call 2 dest-interface IF_ISDN_01
route call 3 dest-interface IF_ISDN_02
route call 4 dest-interface IF_ISDN_03
interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-service SRV_HG
local
service sip-location-service SER_SIP_SER
bind location-service SER_LOC
authentication-service AUTH_SRV
realm patton
username vigili password PYHciCqT3LgQ1iW8rLkpfg== encrypted
location-service SER_LOC
domain 1 192.168.10.160
match-any-domain
identity-group DEFAULT
authentication inbound
authenticate 1 authentication-service AUTH_SRV username vigili
registration inbound
identity vigili inherits DEFAULT
context sip-gateway GW_SIP
interface SIP
transport-protocol udp+tcp 5060
no transport-protocol tls
bind ipaddress ROUTER WAN DHCP
context sip-gateway GW_SIP
no shutdown
port ethernet 0 0
bind interface ROUTER WAN
no shutdown
port bri 0 0
clock auto
encapsulation q921
q921
protocol pmp
permanent-layer2
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_00
port bri 0 0
no shutdown
port bri 0 1
clock auto
encapsulation q921
q921
protocol pmp
permanent-layer2
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_01
port bri 0 1
no shutdown
port bri 0 2
clock auto
encapsulation q921
q921
protocol pmp
permanent-layer2
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_02
port bri 0 2
no shutdown
port bri 0 3
clock auto
encapsulation q921
q921
protocol pmp
permanent-layer2
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_03
port bri 0 3
no shutdown
Thanks in advance
Davide
You have provided no Asterisk related information (Asterisk logs and configuration).
Google fails to find that model number. Is it an ISDN to SIP gateway?
Hi, thaks for replay, exact model is:
SN4131/4BIS8VHP/EUI
4 BRI port connected to 4 ISDN
Asterisk 13.27.1
On asterisk CLI no logs for inbound calls, about me the calls are no routed to the asterisk
P.S:
I use FreePBX
From cli if type sip show peers i seet that is registerd on patton
Check the SIP logs for he registration, paraticularly that the Contact header is correct.
I’l check.
The registration may be right t in one way only or if it is correct it is correct for both? Because if in outgoing the calls works as expected i suppose that the registration is correct.
I cofirm, any logs (full, sip, ecc), are populated only for outbound calls, no trace about incoming calls, for me the problem is before freepbx, patton don’t send calls to freepbx, but the patton conf seems correct.
SIP does not require one to register before making outbound calls, although some systems might only accept calls from peers to whci they would send calls.
Even if they insist on registration, the Contact header in the registration may not contain and address to whcih they can route.
Ok understand, the problem may be an incorrect sip header from patton calls?
The problem may be an incorrect NAT configuration on Asterisk resulting in its supplying a local address when you need a public addrss, or v.v.
I’m trying, i’ve tried with nat yes e no but on each BRI port i can only send calls but i can’t recive anything. Ive enabled sip debug but during the incoming calls (i hear the ring) but on cli nothing apper and i’ve debut set to 7
Don’t stab in the dark. Look a the logs and work out if the wong address is actually being sent. Note nat=yes is deprecated.
yes of curse, i’m using nat=force_rport
I’m in tail on the log but during incoming calls i dont see nothing…
If you have a NAT problem it is unlikely to relate to the nat= setting
Please capture the REGISTER transaction and post it here, being very careful not to lose the distinction between public and private addresses in any redaction.
SIP/2.0 200 OK
Accept: application/sdp, application/dtmf-relay, application/hook-flash, application/QSIG, application/broadsoft, application/vnd.etsi.aoc+xml
Via: SIP/2.0/UDP 192.168.10.160:5060;branch=z9hG4bK0c44fba6;rport=5060;received=192.168.10.160
From: “Unknown” sip:Unknown@192.168.10.160;tag=as0ae9b884
To: sip:192.168.10.64;tag=4145122018
Call-ID: 0cbb3e227f7c49bf502037ab785c6f3d@192.168.10.160:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER
Server: Patton SN4131/4BIS8VHP 00A0BA0FB93E 3.16.0-19083 1.8 M5T SIP Stack/4.2.28.153
Content-Length: 0
Freepbx and patton are in the same lan as te phones
That’s not a complete transaction and it isn’t a REGISTER transaction.
Also, please mark logs as preformatted text, using the </> button.
what do you mean for REGISTER transaction
In asterisk log folder ive only this log
fail2ban.log
full.log
freepbx.log
And the only REGISTER voice is what i posted before adn is in full.log
Please provide the complete sip.conf or pjsip.conf configuration affecting the gateway device.
You said “i seet that is registerd on patton”. To me that indicates that Asterisk must have initiated an, apparently, successful REGISTER transaction. I would like to see that SIP exchange that implemented that.
I cat upload attach as new user, i send it i next 2 posts
sip_additional and sip.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf
[root@freepbx asterisk]# vi sip_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[103]
deny=0.0.0.0/0.0.0.0
secret=1234
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/103
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Ufficio Viabilità <103>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[105]
deny=0.0.0.0/0.0.0.0
secret=1234
"sip_additional.conf" 326L, 5489C
deny=0.0.0.0/0.0.0.0
secret=1234
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/115
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Postazione dietro Co <115>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[vigili]
username=vigili
type=friend
secret=2019sistemi!
qualify=yes
insecure=port,invite
host=192.168.10.64
dtmfmode=rfc2833
context=from-pstn
sendrpid=yes
nat=force_rport
sip.conf
sip_additional.conf sip_custom_post.conf sip_nat.conf sip_notify_custom.conf
sip.conf sip_general_additional.conf sip_notify_additional.conf sip_registrations.conf
sip_custom.conf sip_general_custom.conf sip_notify.conf sip_registrations_custom.conf
[root@freepbx asterisk]# vi sip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications. ;
;--------------------------------------------------------------------------------;
[general]
; These files will all be included in the [general] context
;
#include sip_general_additional.conf
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf
[root@freepbx asterisk]# vi sip_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[103]
deny=0.0.0.0/0.0.0.0
secret=1234
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/103
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Ufficio Viabilità <103>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[105]
deny=0.0.0.0/0.0.0.0
secret=1234
"sip_additional.conf" 326L, 5489C
deny=0.0.0.0/0.0.0.0
secret=1234
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/115
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Postazione dietro Co <115>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[vigili]
username=vigili
type=friend
secret=2019sistemi!
qualify=yes
insecure=port,invite
host=192.168.10.64
dtmfmode=rfc2833
context=from-pstn
sendrpid=yes
nat=force_rport
[root@freepbx asterisk]# vi sip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications. ;
;--------------------------------------------------------------------------------;
[general]
; These files will all be included in the [general] context
;
#include sip_general_additional.conf
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
The registration is outbound, freepbx make a registration on patton
There appears to be no registration involved. I’d say the gateway simply has no idea how to reach the Asterisk box.
The gateway may require Asterisk to register, but there is nothing in the configuration that would achieve that.