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patton 4114 gateways


speck

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having an issue with my patton 4114 gateways

 

problem is this, calls come in, ring twice on my system, and then stop ringing, however they continue to ring for the calling party..

 

originally the pattons were sending calls to anonymous, this was changed as it appeared pbxnsip was rejecting the anonymous... so now they are being sent to the receptionist extension (hunt group consisting of 2 extensions)

 

however the calls are still only ringing on these phones twice, and disappearing.

 

configurations are as follows

 

 

cli version 3.20

clock local default-offset -04:00

dns-client server 8.8.8.8

webserver port 80 language en

sntp-client

sntp-client server primary 172.169.1.xxx port 123 version 4

system hostname "PSTN Gateway A"

 

system

 

ic voice 0

low-bitrate-codec g729

 

profile ppp default

 

profile call-progress-tone defaultDialtone

play 1 1000 450 -6

 

profile call-progress-tone defaultAlertingtone

play 1 1000 450 -13

pause 2 5000

 

profile call-progress-tone defaultBusytone

play 1 300 450 -7

pause 2 300

 

profile call-progress-tone defaultReleasetone

play 1 300 450 -7

pause 2 300

 

profile call-progress-tone defaultCongestiontone

play 1 300 450 -7

pause 2 300

 

profile tone-set default

 

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

 

profile pstn default

 

profile sip default

no autonomous-transitioning

 

profile aaa default

method 1 local

method 2 none

 

context ip router

 

interface IF_IP_LAN

ipaddress 172.169.1.xxx 255.255.255.0

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

 

interface IF_IP_WAN

ipaddress dhcp

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

 

context cs switch

no digit-collection timeout

 

interface sip IF_SIP_1

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_2

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_3

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_4

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface fxo IF_FXO_1

route call dest-interface IF_SIP_1

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_2

route call dest-interface IF_SIP_2

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_3

route call dest-interface IF_SIP_3

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_4

route call dest-interface IF_SIP_4

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

service hunt-group HUNT_FXO

cyclic

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

drop-cause destination-out-of-order

route call 1 dest-interface IF_FXO_1

route call 2 dest-interface IF_FXO_2

route call 3 dest-interface IF_FXO_3

route call 4 dest-interface IF_FXO_4

 

context cs switch

no shutdown

 

context sip-gateway GW_SIP_ALL_LINES

 

interface LAN

bind interface IF_IP_LAN context router port 5060

 

context sip-gateway GW_SIP_ALL_LINES

no shutdown

 

port ethernet 0 0

medium auto

encapsulation ip

bind interface IF_IP_LAN router

no shutdown

 

port fxo 0 0

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_1 switch

no shutdown

 

port fxo 0 1

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_2 switch

no shutdown

 

port fxo 0 2

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_3 switch

no shutdown

 

port fxo 0 3

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_4 switch

no shutdown

 

 

 

 

 

cli version 3.20

clock local default-offset -04:00

dns-client server 8.8.8.8

webserver port 80 language en

sntp-client

sntp-client server primary 172.169.1.xxx port 123 version 4

system hostname "PSTN Gateway B"

 

system

 

ic voice 0

low-bitrate-codec g729

 

profile ppp default

 

profile call-progress-tone defaultDialtone

play 1 1000 450 -6

 

profile call-progress-tone defaultAlertingtone

play 1 1000 450 -13

pause 2 5000

 

profile call-progress-tone defaultBusytone

play 1 300 450 -7

pause 2 300

 

profile call-progress-tone defaultReleasetone

play 1 300 450 -7

pause 2 300

 

profile call-progress-tone defaultCongestiontone

play 1 300 450 -7

pause 2 300

 

profile tone-set default

 

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

 

profile pstn default

 

profile sip default

no autonomous-transitioning

 

profile aaa default

method 1 local

method 2 none

 

context ip router

 

interface IF_IP_LAN

ipaddress 172.169.1.xxx 255.255.255.0

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

 

interface IF_IP_WAN

ipaddress dhcp

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

 

context cs switch

no digit-collection timeout

 

interface sip IF_SIP_1

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_2

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_3

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface sip IF_SIP_4

bind context sip-gateway GW_SIP_ALL_LINES

route call dest-service HUNT_FXO

remote 172.169.1.xxx 5060

early-connect

early-disconnect

address-translation outgoing-call to-header user-part fix 801 host-part remote

 

interface fxo IF_FXO_1

route call dest-interface IF_SIP_1

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_2

route call dest-interface IF_SIP_2

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_3

route call dest-interface IF_SIP_3

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

interface fxo IF_FXO_4

route call dest-interface IF_SIP_4

loop-break-duration min 200 max 1000

disconnect-signal loop-break

disconnect-signal busy-tone

ring-number on-caller-id

dial-after timeout 1

mute-dialing

 

service hunt-group HUNT_FXO

cyclic

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

drop-cause destination-out-of-order

route call 1 dest-interface IF_FXO_1

route call 2 dest-interface IF_FXO_2

route call 3 dest-interface IF_FXO_3

route call 4 dest-interface IF_FXO_4

 

context cs switch

no shutdown

 

context sip-gateway GW_SIP_ALL_LINES

 

interface LAN

bind interface IF_IP_LAN context router port 5060

 

context sip-gateway GW_SIP_ALL_LINES

no shutdown

 

port ethernet 0 0

medium auto

encapsulation ip

bind interface IF_IP_LAN router

no shutdown

 

port fxo 0 0

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_1 switch

no shutdown

 

port fxo 0 1

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_2 switch

no shutdown

 

port fxo 0 2

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_3 switch

no shutdown

 

port fxo 0 3

use profile fxo us

caller-id format bell

encapsulation cc-fxo

bind interface IF_FXO_4 switch

no shutdown

 

 

 

any ideas??

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keep in mind this system ran for over a year, with 0 repeat Z E R O problems, with a grandstream gateway, until that gateway crapped out on me (started locking up and not allowing calls in or out without a power cycle)

 

once we switched to the patton gateways, this is when the problem started..

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Hi,

 

Can you enable SIP Logging and post the logs here?

patton went through debug logs of the gateways.. apparently pbxnsip is not sending a 200 ok message to the gateways, or if it is the gateways are not receibing it, therefore after 30 seconds the pattons send a cancel message to pbxnsip

 

sorry for mistakes typing rapidly on Iphone haha

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patton went through debug logs of the gateways.. apparently pbxnsip is not sending a 200 ok message to the gateways, or if it is the gateways are not receibing it, therefore after 30 seconds the pattons send a cancel message to pbxnsip

 

sorry for mistakes typing rapidly on Iphone haha

 

 

ok well my vendor got it working, odly enough we had to remove encryption, and update some firmware but its working great now..

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