Patton 4524 Unable to receive incoming call

Hello All,
I configured my patton 4524 - 4 FXO and I’m able to made outbond calls, but when I try to call a internal by mobile I get this error
(callid 835f2894175557fa) - No matching endpoint found
My trunk is configured in “friend” mode:
I try to make a pjsip trunk and I receive the call in extension 301 (endpoint of inbound route) but I’m cannot hear the voice.
My question is : Is The Trunk type 'friend ’ enough to receive and send call by patton?
Any Idea to fix my issue?



R6.9 2017-05-04 H323 SIP FXS FXO



Generated configuration file


cli version 3.20
clock local default-offset +01:00
dns-client server
webserver port 80 language en
sntp-client server primary port 123 version 4
system hostname SN4524


ic voice 0

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

profile aaa default
method 1 local
method 2 none

context ip router

interface LAN
no napt-inside

context ip router
route 0

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

routing-table called-e164 RT_IN_FXO1
route default dest-interface IF_SIP1

interface sip IF_SIP1
bind context sip-gateway GW_SIP_1
route call dest-service SER_HG_FXO
remote 5060

interface fxo IF_FXO1
route call dest-table RT_IN_FXO1
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1

service hunt-group SER_HG_FXO
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_FXO1

context cs switch
no shutdown

authentication-service AUTH_SVC
realm 1 from-pstn
username PattonUser1 password 36ocYTYpKxk= encrypted

location-service LOCATION_SVC_1
domain 1

identity PattonUser1

authentication inbound

context sip-gateway GW_SIP_1

interface IF_GW_SIP_1
bind interface LAN context router port 5060

context sip-gateway GW_SIP_1
bind location-service LOCATION_SVC_1
no answer-untrusted-hosts
no shutdown

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

port ethernet 0 1
medium negotiated 10 half

port fxo 0 0
encapsulation cc-fxo

port fxo 0 1
encapsulation cc-fxo
bind interface IF_FXO1 switch
no shutdown

port fxo 0 2

port fxo 0 3

For the chan_sip case, assuming that Bind Port has the default value of 5160, in the Patton change
remote 5060
remote 5160

For pjsip, check that in Asterisk SIP settings, Local Networks and External Address are correctly set.

If no luck, at the Asterisk command prompt, type
pjsip set logger on
sip set debug on
make a failing test call, paste the relevant section of the Asterisk log at and post the link here.

1 Like

This occurs when you are trying to reach an extension in the “wrong” SIP channel driver. For example, you are receiving the call on the Chan-SIP port, but the extension is setup in PJ-SIP, or vice versa. Make sure your ports associated with the channel drivers (5060 is PJ-SIP and 5160 is Chan-SIP by default) are the ones you are expecting.

Voice problems like this are typically RTP problems, which are on UDP ports 10000-20000. This is separate (largely) from the SIP problem you are describing. I don’t see where, in your configuration, you specify the RTP channel range, but that would be the thing I’d probably start looking for next. If the RTP channel space isn’t modifiable, you can tune the PBX to use the range that the Patton is using.

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