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- Aug 8, 2007
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Hello,
i think 3cx is a great VOIP solution and we are thinking of buying the enterprise solution. But i have some problems with ISDN. I am new in VOIP and i hope you can help me with configure my Patton Smartnode 4552.
I am from Germany and have an ISDN S0 Port, which is connected to the Patton Gateway.
My 3cx works internal fine, and all phones can call internal.
Now i want to connect my ISDN Port with with the Patton VOIP Gateway. I have configured my patton and i used this instruction:
Configuring a Patton Smartnode ISDN for 3CX Phone System:
http://www.3cx.com/voip-gateways/smartnode-isdn.html
The 2 PSTN Lines register to 3cx now and all incoming calls, let the phones ring. But when i answer the call, i can hear anyone, and the other way around.
One more problem is, that i can't call out to someone. When i want to make a outgoing call nothing happens on softphone and the logfile from 3cx says:
[CM103002] Call(43): Incoming call from 102 (Ext.102) to sip:[email protected]
1 Minute after i cancel the Call because i will not connect:
CM104007] Call(43): Call from Ext.102 to 008073916313 has been terminated by Ext.102; cause: CANCEL; from IP:192.168.0.13
Here is my context cs:
3CX = 192.168.178.41
Patton = 192.168.178.1
Can anybody give me a hint where the problem is?
Thank you very much, nice greetings from munich
Dida
(Sorry for my bad english :? )
i think 3cx is a great VOIP solution and we are thinking of buying the enterprise solution. But i have some problems with ISDN. I am new in VOIP and i hope you can help me with configure my Patton Smartnode 4552.
I am from Germany and have an ISDN S0 Port, which is connected to the Patton Gateway.
My 3cx works internal fine, and all phones can call internal.
Now i want to connect my ISDN Port with with the Patton VOIP Gateway. I have configured my patton and i used this instruction:
Configuring a Patton Smartnode ISDN for 3CX Phone System:
http://www.3cx.com/voip-gateways/smartnode-isdn.html
The 2 PSTN Lines register to 3cx now and all incoming calls, let the phones ring. But when i answer the call, i can hear anyone, and the other way around.
One more problem is, that i can't call out to someone. When i want to make a outgoing call nothing happens on softphone and the logfile from 3cx says:
[CM103002] Call(43): Incoming call from 102 (Ext.102) to sip:[email protected]
1 Minute after i cancel the Call because i will not connect:
CM104007] Call(43): Call from Ext.102 to 008073916313 has been terminated by Ext.102; cause: CANCEL; from IP:192.168.0.13
Here is my context cs:
3CX = 192.168.178.41
Patton = 192.168.178.1
Code:
context cs switch
national-prefix 0
international-prefix 00
routing-table called-e164 RT_SPEED_DIAL
route T dest-table RT_NR_BLOCKING MT_SPEED_DIAL
routing-table called-e164 RT_NR_BLOCKING
route default dest-table RT_CALL_ROUTING MT_SET_CNPN
routing-table called-e164 RT_CALL_ROUTING
route default dest-service SER_HG_PSTN_FALLBACK
routing-table called-e164 RT_INCOMING
route default dest-interface IF_S0_01 MT_NR_TRANSLATION
mapping-table called-e164 to called-e164 MT_SPEED_DIAL
mapping-table calling-e164 to calling-e164 MT_SET_CNPN
mapping-table called-e164 to called-e164 MT_NR_TRANSLATION
interface isdn IF_S0_00
route call dest-table RT_INCOMING
interface isdn IF_S0_01
route call dest-table RT_SPEED_DIAL
use profile tone-set Europe
isdn-date-time
interface isdn IF_DEV0
interface isdn ISDN0
route call dest-interface SIPIF0
interface isdn ISDN1
route call dest-interface SIPIF1
interface sip IF_SIP_SERVICE
bind gateway GW_SIP
service default
route call dest-table RT_INCOMING
use profile voip VOIP
interface sip SIPIF0
bind gateway SIPGW0
service default
route call dest-interface ISDN0
remote-party-id called-party
address-translation outgoing-call to-header user-part fix 10001 host-part interface
address-translation incoming-call called-e164 request-uri
interface sip SIPIF1
bind gateway SIPGW1
service default
route call dest-interface ISDN1
remote-party-id called-party
address-translation outgoing-call to-header user-part fix 10002 host-part interface
address-translation incoming-call called-e164 request-uri
service hunt-group SER_HG_PSTN_FALLBACK
timeout 6
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 no-route-to-destination
route call 1 dest-interface IF_S0_00
route call 2 dest-interface IF_SIP_SERVICE
context cs switch
no shutdown
gateway sip GW_SIP
bind interface IF_IP_WAN router
service default
gateway sip GW_SIP
no shutdown
gateway sip SIPGW0
call-signaling-port 5062
bind interface IF_IP_LAN router
service default
domain 192.168.178.41
authentication 10001 password dZ8edXkjFnM= encrypted default
default-server 192.168.178.41 loose-router
registration-lifetime 120
registrar 192.168.178.41
user 10001
gateway sip SIPGW0
no shutdown
gateway sip SIPGW1
call-signaling-port 5064
bind interface IF_IP_LAN router
service default
domain 192.168.178.41
authentication 10002 password iAaKwKfxX4Q= encrypted default
default-server 192.168.178.41 loose-router
registration-lifetime 120
registrar 192.168.178.41
user 10002
gateway sip SIPGW1
no shutdown
Can anybody give me a hint where the problem is?
Thank you very much, nice greetings from munich
Dida
(Sorry for my bad english :? )