Chopped Voice

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pangasewa

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Apr 4, 2008
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Hi, Since we started using 3cx we have been having issues such as sometime we are getting voice being chopped out from internal calling but also if inbound call coming in. We have made some changes to our configuration after Patton team suggested what we need to try but the problem still exist. it happens more when we connect atleast more than one PSTN line to the Patton Gateway.

We are using 3cx free version and we have patton 4114 and on the phones we have a mix of snom320 and grandstreams 2000. We are not using any Voip provide, its just normal PSTN line from the bell company.

we have taken out the codec g729 and leave G711 per recommendation from Patton team.

please send us you comments that may help us solve this issue, we have spent quite number of days without an answer and we are thinking of pulling out from this project.

here is our configuration

webserver port 80 language en

system

ic voice 0

profile ppp default

profile call-progress-tone US_Dialtone
flush-play-list
play 1 1000 350 -13 440 -13

profile call-progress-tone US_Alertingtone
flush-play-list
play 1 1000 440 -19 480 -19
pause 2 3000

profile call-progress-tone US_Busytone
flush-play-list
play 1 500 480 -24 620 -24
pause 2 500

profile call-progress-tone US_Releasetone
flush-play-list
play 1 250 480 -24 620 -24
pause 2 250

profile call-progress-tone US_Congestiontone
flush-play-list
play 1 250 480 -24 620 -24
pause 2 250

profile tone-set US_default
map call-progress-tone dial-tone US_Dialtone
map call-progress-tone ringback-tone US_Alertingtone
map call-progress-tone busy-tone US_Busytone
map call-progress-tone release-tone US_Releasetone
map call-progress-tone congestion-tone US_Congestiontone

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

profile voip fax_enabled
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k 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 eth0
ipaddress 10.7.1.111 255.255.255.0

context cs switch
no digit-collection timeout

interface sip sipif0
bind gateway sipgw0
service default
route call dest-service fxohuntgroup
address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none

interface sip sipif1
bind gateway sipgw1
service default
route call dest-service fxohuntgroup
address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none

interface sip sipif2
bind gateway sipgw2
service default
route call dest-service fxohuntgroup
address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none

interface sip sipif3
bind gateway sipgw3
service default
route call dest-service fxohuntgroup
address-translation outgoing-call request-uri user-part fix 10003 host-part to-header target-param none

interface fxo fxoif0
route call dest-interface sipif0
disconnect-signal battery-reversal
disconnect-signal busy-tone
dial-after timeout 1
use profile tone-set US_default

interface fxo fxoif1
route call dest-interface sipif1
disconnect-signal battery-reversal
disconnect-signal busy-tone
dial-after timeout 1
use profile tone-set US_default

interface fxo fxoif2
route call dest-interface sipif2
disconnect-signal battery-reversal
disconnect-signal busy-tone
dial-after timeout 1
use profile tone-set US_default

interface fxo fxoif3
route call dest-interface sipif3
disconnect-signal battery-reversal
disconnect-signal busy-tone
dial-after timeout 1
use profile tone-set US_default

service hunt-group fxohuntgroup
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 user-busy
route call 1 dest-interface fxoif0
route call 2 dest-interface fxoif1
route call 3 dest-interface fxoif2
route call 4 dest-interface fxoif3

context cs switch
no shutdown

gateway sip sipgw0
bind interface eth0 router

service default
domain 10.7.1.127
authentication 10000 password 10000 default
default-server 10.7.1.127 loose-router
registrar 10.7.1.127
user 10000

gateway sip sipgw0
no shutdown

gateway sip sipgw1
bind interface eth0 router

service default
domain 10.7.1.127
authentication 10001 password 10001 default
default-server 10.7.1.127 loose-router
registrar 10.7.1.127
user 10001

gateway sip sipgw1
no shutdown

gateway sip sipgw2
bind interface eth0 router

service default
domain 10.7.1.127
authentication 10002 password 10002 default
default-server 10.7.1.127 loose-router
registrar 10.7.1.127
user 10002

gateway sip sipgw2
no shutdown

gateway sip sipgw3
bind interface eth0 router

service default
domain 10.7.1.127
authentication 10003 password 10003 default
default-server 10.7.1.127 loose-router
registrar 10.7.1.127
user 10003

gateway sip sipgw3
no shutdown

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

port fxo 0 0
use profile fxo us
encapsulation cc-fxo
bind interface fxoif0 switch
no shutdown

port fxo 0 1
use profile fxo us
encapsulation cc-fxo
bind interface fxoif1 switch
no shutdown

port fxo 0 2
use profile fxo us
encapsulation cc-fxo
bind interface fxoif2 switch
no shutdown

port fxo 0 3
use profile fxo us
encapsulation cc-fxo
bind interface fxoif3 switch
no shutdown

thanks
 
We have similar issues but not as many as before this change-
Made sure that all Gateways and other phones use G711u first. As the Patton gateways only have 3 codecs available, our first in the list is G711u then G711a then G729 in that order, I didn't remove the G729 just moved it to the bottom. The phones also are PCMU first which I understand is G711U. In 3CX the codecs seem to be able to be set under the provisioning tab, but because Grandstreams are not fully supported anymore, I set this in the phones themselves. I figure that the least searching in a list of matching codecs may ease the connections. G729 has been shown at our place to have poor quality and it seems after about 3 to 5 minutes becomes worse. I am still searching for the perfect cure to this problem but this has definitely helped.
 
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