Can't make outbound calls from extension to PSTN

[brianfpmoab]

Hi,

I have 3CX v3.0.1699.0 installed and configured for 2 extensions and 7PSTN lines via 2 PSTN gateways. I can successfully make calls from an outside line (cell phone, home phone etc) to any of the lines connected to either of the gateways and am passed into the PBX just fine. I can call from extension to extension just fine. I have an outbound rule for each gateway that routes calls to each gateway and strips the first digit off. The problem is that when I make a call from either extension I get the following in the log.

07:34:05.140 CallConf::Rejected Call (C:11) is rejected: Not Found
07:34:05.140 StratInOut:nCancel Call from Ext.103 to 2594384 has been terminated; reason: Not Found
07:34:02.531 CallConf:nIncoming Incoming call from Ext.103 to sip:[email protected]

The devices are on the same network subnet. The phone, gateway, and PBX are connected to the same switch. I do not have a firewall running on the PBX.

My question is: Is this a problem in the configuration of the PBX, or in one of the hardware devices (ATA adaptor or Gateway)?

I have only added 2 extensions for testing . When the system is up and running I will have 12-15 extensions.

Any help would be appreciated.

Thanks

Brian

 

[NigelG]

RE: Problem

Hi Brian,

In the help file under troubleshooting, the following seems to match your problem which indicates that your PSTN Gateway is unable to dial the requested number:

b. Failed call: VoIP provider can not dial specified number

 

[archie]

Hi Brian,

Nigel is right. Your device rejects numbers you've dialled by some reason. Please, tell us what exactly device do you have, and what are your 3CX settings (including Advanced options) for that device. Than we probably can suggest how to set things up.

 

[brianfpmoab]

Nigel and Archie,

Here are the settings in 3CX for the gateway.
VOIP Gateway
Name: Moab
Gateway Hostname or IP: xx.xx.xx.131
Gateway Port: 5060
Advanced Options
Config template: Generic
Device is external: unchecked
Bind to Media Server: unchecked
Use IP in 'Contact' to recognize source: Checked
Caller-ID is in

 

[brianfpmoab]

Nigel and Archie,

Thank you for pointing me in the right direction. I was able to get everything to work.

Just FYI for anyone else using an AudioCodes gateway. I was not able to make outbound calls from an extension with the errors mentioned in the first post. The solution was in the AudioCodes device in the IP to hunt group routing table. I ended up making the dest prefix, source prefix and source ip all * and the hunt group ID of 1 which I had setup earlier.

Thanks again!

Brian

 

[dyntech]

Brian,
I have the same device and can call out but will not accept incoming calls. Can you either send me or post your ini file for MP-118?


Thanks
Mike

Got It working finally!!!

For anyone else with the same hardware here is the ini file:

;**************
;** Ini File **
;**************

;Board: MP-118 FXO
;Serial Number: 761897
;Slot Number: 1
;Software Version: 5.00A.031.003
;DSP Software Version: 204IM => 209.16
;Board IP Address: 192.168.80.15
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.168.80.1
;Ram size: 32M Flash size: 8M
;Num DSPs: 2 Num DSP channels: 8
;Profile: NONE
;-----------------------------------------


[SYSTEM Params]

NTPServerIP = 192.168.80.10
TelnetServerEnable = 1
ActivityListToLog = 'pvc', 'dr', 'fb', 'swu', 'spc'

[BSP Params]

PCMLawSelect = 3
BaseUDPPort = 6000
LocalOAMIPAddress = 192.168.80.15
LocalOAMSubnetMask = 255.255.255.0
LocalOAMDefaultGW = 192.168.80.1
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

[ATM Params]


[Analog Params]

FXOLoopCharacteristicsFilename = 'MP1xx12-1-16khz-fxo.dat'
FlashHookPeriod = 500

[ControlProtocols Params]

MGCPDTMFDetectionPoint = 0
AdminStateLockControl = 0

[MGCP Params]


[MEGACO Params]

EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0

[PSTN Params]


[SS7 Params]


[Voice Engine Params]

EnableContinuityTones = 1
IdlePCMPattern = 255
L1L1ComplexRxUDPPort = 0
L1L1ComplexTxUDPPort = 0
EnableSilenceCompression = 0
EnableEchoCanceller = 1
ECHybridLoss = 1
VoiceVolume = 20
InputGain = -12
DTMFVolume = -20
DTMFTransportType = 3
MFTransportType = 3
CallerIDTransportType = 1
V22ModemTransportType = 0
V23ModemTransportType = 0
V32ModemTransportType = 0
V34ModemTransportType = 0
FaxRelayECMEnable = 0
BasicRTPPacketInterval = 0
CNGDetectorMode = 1
DJBufMinDelay = 20
DJBufOptFactor = 13
RTPRedundancyDepth = 0
RTPPackingFactor = 1
RTPDirectionControl = 0
RFC2833PayloadType = 101
RFC2833TxPayloadType = 96
RFC2833RxPayloadType = 96
RFC2198PayloadType = 104
FaxBypassPayloadType = 102
EnableStandardSIDPayloadType = 0
EnableAnswerDetector = 1
AnswerDetectorActivityDelay = 0
AnswerDetectorSilenceTime = 10
AnswerDetectorRedirection = 0
AnswerDetectorSensitivity = 0
AnalogSignalTransportType = 1
CallProgressTonesFilename = 'usa_tones_12.dat'

[WEB Params]

LogoWidth = '339'

[SIP Params]

ENABLECALLERID = 1
MAXDIGITS = 10
ALWAYSUSEROUTETABLE = 0
PLAYRBTONE2IP = 1
REGISTRATIONTIME = 3600
USERADLOG = 1
ISPROXYUSED = 0
ISREGISTERNEEDED = 1
AUTHENTICATIONMODE = 0
ISWAITFORDIALTONE = 1
ISTWOSTAGEDIAL = 0
ENABLECURRENTDISCONNECT = 1
RADDEBLEVEL = 2
MAXACTIVECALLS = 4
CHANNELSELECTMODE = 2
RADLOGOUTPUT = 1
GWDEBUGLEVEL = 6
ENABLEPROXYKEEPALIVE = 0
ENABLERPIHEADER = 1
ENABLEEARLYMEDIA = 1
DEFAULTNUMBER = '101'
PROXYNAME = '3cx'
REGISTRARIP = '192.168.80.11'
SIPGATEWAYNAME = '192.168.80.15'
USERNAME = ''
CNONCE = '0a123bcf'
PASSWORD = ''
ISFALLBACKUSED = 0
ALWAYSSENDTOPROXY = 1
ISPROXYHOTSWAP = 0
PROXYKEEPALIVETIME = 60
PROXYREDUNDANCYMODE = 0
PROXYHOTSWAPRTX = 3
SENDINVITETOPROXY = 1
REGISTRATIONRETRYTIME = 30
USEGATEWAYNAMEFOROPTIONS = 0
ADDTON2RPI = 0
PREFERROUTETABLE = 0
USESOURCENUMBERASDISPLAYNAME = 1
SUBSCRIPTIONMODE = 1
GWREGISTRATIONNAME = '192.168.80.15'
REGISTRARNAME = '192.168.80.11'
REGISTRATIONTIMEDIVIDER = 50
RINGSBEFORECALLERID = 2
REMOTEBASEUDPPORT = 0
COMFORTNOISENEGOTIATION = 1
CODERNAME = g711Ulaw64k,20,0,0,0
CALLERDISPLAYINFO7 = Front Door,0
PREFIX = *,192.168.80.11,*,1,255
PSTNPREFIX = *,1,*,*,0
TARGETOFCHANNEL4 = 999,0
TARGETOFCHANNEL5 = 999,0
TARGETOFCHANNEL6 = 999,0
TARGETOFCHANNEL7 = 998,1
TARGETOFCHANNEL0 = 10000,1
TARGETOFCHANNEL1 = 10001,1
TARGETOFCHANNEL2 = 10002,1
TARGETOFCHANNEL3 = 10003,1
TRUNKGROUP_1 = 1-1,10000,0
TRUNKGROUP_1 = 2-2,10001,0
TRUNKGROUP_1 = 3-3,10002,0
TRUNKGROUP_1 = 4-4,10003,0
PROXYIP = 192.168.80.11
AUTHENTICATION_0 = 10000,10000
AUTHENTICATION_1 = 10001,10001
AUTHENTICATION_2 = 10002,10002
AUTHENTICATION_3 = 10003,10003
TRUNKGROUPSETTINGS = 1,1,0,$$
TXDTMFOPTION = 4
;TelProfile: ProfileName, Preference, CodersGroupID, IsFaxUsed, DJBufMinDelay, JBufOptFactor, IPDiffServ, SigIPDiffServ, DtmfVolume, InputGain, VoiceVolume, EnableReversePolarity, EnableCurrentDisconnect, EnableDigitDelivery, ECE, MWIAnalog, MWIDisplay, FlashHookPeriod, EnableEarlyMedia, ProgressIndicator2IP
TELPROFILE_1 = Default Tel Profile,1,0,0,10,10,46,40,-6,-6,10,0,1,0,1,0,0,400,1,-1
ENABLECALLERID_0 = 1
ENABLECALLERID_1 = 1
ENABLECALLERID_2 = 1
ENABLECALLERID_3 = 1
ENABLECALLERID_4 = 1
ENABLECALLERID_5 = 1
ENABLECALLERID_6 = 1
ENABLECALLERID_7 = 1

[VXML Params]


[IPsec Params]


[Audio Staging Params]

DisableSNMP = 1

;
; *** TABLE DspTemplates ***
; This table contains hidden elements and will not be exposed.
; This table exists on board and will be saved during restarts
;

 

[brianfpmoab]

Mike,

Here is my ini file.

;**************
;** Ini File **
;**************

;Board: MP-118 FXO
;Serial Number: 710115
;Slot Number: 1
;Software Version: 4.80A.014.006
;Board IP Address: xx.xx.xx.131
;Board Subnet Mask: 255.255.255.128
;Board Default Gateway: xx.xx.xx.129
;Ram size: 32M Flash size: 8M
;Num DSPs: 2 Num DSP channels: 8
;Profile: NONE
;------------------------------


[SYSTEM Params]

DisableSNMP = 1
DNSPriServerIP = xx.xx.x.2
SyslogServerIP = 10.1.1.89
VoiceMenuPassword = 'disable'

[BSP Params]

PCMLawSelect = 3
LocalOAMIPAddress = xx.xx.xx.131
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

[ATM Params]


[Analog Params]

CallProgressTonesFilename = 'usa_tones_12.dat'

[ControlProtocols Params]


[MGCP Params]


[MEGACO Params]

EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0

[SS7 Params]


[Voice Engine Params]

IdlePCMPattern = 85
EnableSilenceCompression = 1
VoiceVolume = 1
CallerIDTransportType = 1
RFC2833PayloadType = 101

[WEB Params]

LogoWidth = '339'

[SIP Params]

ENABLECALLERID = 1
MAXDIGITS = 10
TIMEBETWEENDIGITS = 6
USERADLOG = 1
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
ISTWOSTAGEDIAL = 0
ENABLEHOLD = 0
RADDEBLEVEL = 2
CHANNELSELECTMODE = 2
RADLOGOUTPUT = 1
GWDEBUGLEVEL = 6
PROXYNAME = 'xx.xx.xx.138'
REGISTRARIP = 'xx.xx.xx.138'
SIPGATEWAYNAME = 'Moab'
ALWAYSSENDTOPROXY = 1
SENDINVITETOPROXY = 1
GWREGISTRATIONNAME = 'Moab'
REGISTRARNAME = 'xx.xx.xx.138'
RINGSBEFORECALLERID = 0
CODERNAME = g711Ulaw64k,20,0,$$,0
CODERNAME = g711Alaw64k,20,0,$$,0
CODERNAME = g7231,30,0,$$,0
CODERNAME_1 = g711Ulaw64k,20,0,$$,0
CODERNAME_1 = g711Alaw64k,20,0,$$,0
CODERNAME_1 = g729,20,0,$$,0
CALLERDISPLAYINFO0 = HOTNC,0
CALLERDISPLAYINFO1 = HOTNC,0
CALLERDISPLAYINFO2 = HOTNC,0
CALLERDISPLAYINFO3 = HOTNC,0
CALLERDISPLAYINFO4 = HOTNC,0
PREFIX = *,xx.xx.xx.138,*,1,255
PSTNPREFIX = *,1,*,*,1
PSTNPREFIX = *,2,*,*,1
TARGETOFCHANNEL0 = 4352594857,1
TARGETOFCHANNEL1 = 4352594019,1
TARGETOFCHANNEL2 = 4352596677,1
TARGETOFCHANNEL4 = 4352598168,1
TARGETOFCHANNEL3 = 4352594384,1
TRUNKGROUP_1 = 1-1,10000,0
TRUNKGROUP_1 = 2-2,10001,0
TRUNKGROUP_1 = 3-3,10002,0
TRUNKGROUP_1 = 4-4,10003,0
TRUNKGROUP_1 = 5-5,10004,0
PROXYIP = xx.xx.xx.138
AUTHENTICATION_0 = 4352594857,4352594857
AUTHENTICATION_1 = 4352594019,4352594019
AUTHENTICATION_2 = 4352596677,4352596677
AUTHENTICATION_4 = 4352598168,4352598168
AUTHENTICATION_3 = 4352594384,4352594384
TRUNKGROUPSETTINGS = 1,3
TRUNKGROUPSETTINGS = 2,1
ENABLECALLERID_0 = 1
ENABLECALLERID_1 = 1
ENABLECALLERID_2 = 1
ENABLECALLERID_3 = 1
ENABLECALLERID_4 = 1

[VXML Params]


[IPsec Params]


[Audio Staging Params]


[PSTN-SDH Params]



Brian

 

[Anonymous]

With these configs, would there be a need to have this consolidated under a similar section as or the VOIP providers?

H.