Patton 4112 pjsip Unable to receive incoming calls

I first configured a Patton SN4112 FXO Gateway using Chan_SIP to make a receive incoming calls and it works flawlessly.

I then changed the SIP server ports to 5060 (from 5160) on the SN4112 and created a PJSIP trunk in FreePBX (v14.0.13.40). I confirmed that FreePBX is using these ports and that both channels are available. I have tried the new trunk and outgoing calls work fine but I cannot receive incoming calls at all. I get nothing in the logs, no error messages nothing…it is as if FreePBX isn’t listening, I just get ringing tone on the external phone.

If I change the channel in the SN4112 SIP back to 5160, I correctly get an error message stating the number is not available which I would expect since I am now using the wrong channel.

I have tried turning off the firewall, disabling authentication etc but that makes no difference.

Any ideas of where I start - maybe I have the trunk configured incorrectly?

The config is below:

SN4112>>>>>>>>>>>>>>>

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

SN4112/JO/EUI

R6.11 2019-07-02 H323 SIP FXS FXO

2020-10-06T16:10:29

SN/00A0BA0B508C

Generated configuration file

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

cli version 3.20
administrator sonusfaber password yZglD7e6PLGtBuqRM97rYg== encrypted
clock local default-offset +01:00
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.1.254 port 123 version 4
sntp-client local-clock-offset

system

ic voice 0

profile ppp default

profile call-progress-tone UK_Dialtone
play 1 5000 350 -13 440 -13

profile call-progress-tone UK_Alertingtone
play 1 400 400 0 450 0
pause 2 200
play 3 400 400 0 450 0
pause 4 2000

profile call-progress-tone UK_Busytone
play 1 375 400 0
pause 2 375

profile call-progress-tone UK_CongestionTone
play 1 400 400 0
pause 2 350
play 3 225 400 0
pause 4 525

profile call-progress-tone UK_WaitingTone
play 1 100 400 0
pause 2 5000

profile call-progress-tone UK_ReleaseTone
play 1 1500 400 0

profile tone-set default
map call-progress-tone dial-tone UK_Dialtone
map call-progress-tone ringback-tone UK_Alertingtone
map call-progress-tone busy-tone UK_Busytone
map call-progress-tone waiting-tone UK_WaitingTone
map call-progress-tone release-tone UK_ReleaseTone
map call-progress-tone congestion-tone UK_CongestionTone

profile tone-set UK
map call-progress-tone dial-tone UK_Dialtone
map call-progress-tone ringback-tone UK_Alertingtone
map call-progress-tone busy-tone UK_Busytone
map call-progress-tone waiting-tone UK_WaitingTone
map call-progress-tone release-tone UK_ReleaseTone
map call-progress-tone congestion-tone UK_CongestionTone

profile voip default
codec 1 g711ulaw64k rx-length 20 tx-length 20
codec 2 g711alaw64k rx-length 20 tx-length 20

profile pstn default
output-gain 3
input-gain 3

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface eth0
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 0.0.0.0 0.0.0.0 192.168.1.254 0

context cs switch

interface sip IF_SIP
bind context sip-gateway GW_SIP_ALL_EXTENSIONS
route call dest-service HG_TO_FXO
remote 192.168.1.118 5060
early-disconnect
address-translation outgoing-call request-uri user-part fix 01946862776 host-part to-header target-param none
trust remote

interface fxo IF_FXO_00
route call dest-interface IF_SIP
loop-break-duration min 200 max 1000
disconnect-signal loop-break
ring-number on-caller-id
dial-after timeout 2
mute-dialing
caller-id format bell

service hunt-group HG_TO_FXO
drop-cause destination-out-of-order
drop-cause user-busy
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_FXO_00

context cs switch
no shutdown

authentication-service AS_ALL_EXTENSIONS
username Patton4112 password necUs1eM+wPTb3ML7e9u+wYBYFYAQ/Nx encrypted

location-service LS_ALL_LINES
domain 1 192.168.1.118
match-any-domain

identity-group default

call inbound

identity 01946862776 inherits default

authentication outbound
  authenticate 1 authentication-service AS_ALL_EXTENSIONS username Patton4112

authentication inbound
  authenticate 1 authentication-service AS_ALL_EXTENSIONS username Patton4112

registration inbound
  contact 192.168.1.118 5060 switch IF_SIP priority 1000
  lifetime default 3600 min 60 max 7200

context sip-gateway GW_SIP_ALL_EXTENSIONS

interface IF_GW_SIP
bind interface eth0 context router port 5060

context sip-gateway GW_SIP_ALL_EXTENSIONS
bind location-service LS_ALL_LINES
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface eth0 router
no shutdown

port fxo 0 0
use profile fxo gb
encapsulation cc-fxo
bind interface IF_FXO_00 switch
no shutdown

port fxo 0 1
shutdown
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

UPDATE:
Got to the bottom of this. Doing a “pjsip show transports” at the Cli, showed the pjsip transport on 5061, despite configuration showing 5060. So I rebooted Freepbx to see how it came back up and it came up on 5060.

I’m not sure where the corruption came from but it is working now.

Port binding changes require an Asterisk restart. This requirement is displayed when you submit the change.

Yes thanks for that, I did notice that it does. Though I wasn’t aware the ports had been changed at all since they were still showing their defaults.

Thanks for the response.

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