Patton 4112 pjsip Unable to receive incoming calls

siptrunk
Tags: #<Tag:0x00007f701f8c8ea8>

(Ian) #1

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
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<


(Ian) #2

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.


(Lorne Gaetz) #3

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


(Ian) #4

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.