Hi All,
I am a first timer in this forum.
I have an Asterisk Server connected to an Audiocodes MP104 FXO gateway. I am able to receive calls from my PSTN lines very well.
My problem however, is outoing calls via PSTN
I am pasting an exerpt from the CLI when I try to dial out:
– Executing Set(“SIP/106-0a1023a8”, “CALLERID(number)=914224380022”) in new stack
– Executing Dial(“SIP/106-0a1023a8”, “SIP/pstn-out/6535596”) in new stack
– Called pstn-out/6535596
– Got SIP response 603 “Decline” back from 192.168.1.125
– SIP/pstn-out-0a1078e8 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing Congestion(“SIP/106-0a1023a8”, “”) in new stack
== Spawn extension (internal, 6535596, 3) exited non-zero on ‘SIP/106-0a1023a8’
– Executing Set(“SIP/106-0a1023a8”, “CALLERID(number)=914224380022”) in new stack
– Executing Dial(“SIP/106-0a1023a8”, “SIP/pstn-out/6535596”) in new stack
– Called pstn-out/6535596
– Got SIP response 603 “Decline” back from 192.168.1.125
– SIP/pstn-out-0a1078e8 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing Congestion(“SIP/106-0a1023a8”, “”) in new stack
== Spawn extension (internal, 6535596, 3) exited non-zero on ‘SIP/106-0a1023a8’
My extensions.conf with respect to PSTN Dial-out is as under:
[ob-pstn]
exten => _NXXXXXX,1,set(CALLERID(number)=914224380022)
exten => _NXXXXXX,2,Dial(SIP/pstn-out/${EXTEN})
exten => _NXXXXXX,3,Congestion()
exten => _NXXXXXX,n+101,Congestion()
From my understanding, there is some setting in the AudioCodes which is refusing SIP to PSTN calls …
My BOARD.ini is as under:
– START FILE –
;Board: MP-104 FXO
;Serial Number: 546265
;Slot Number: 1
;Software Version: 4.60A.016.003
;Board IP Address: 192.168.1.125
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.168.1.1
;Ram size: 16M Flash size: 4M
;Num DSPs: 1 Num DSP channels: 4
;Profile: NONE
;------------------------------
[SYSTEM Params]
DisableSNMP = 1
SyslogServerIP = 10.1.1.89
VXMLFIleName = ‘’
[BSP Params]
[ATM Params]
[Analog Params]
LifeLineType = 1
MinFlashHookTime = 25
FXOLoopCharacteristicsFilename = 'MP1xx12-1-16khz-fxo.dat’
FarEndDisconnectSilencePeriod = 60
CallProgressTonesFilename = 'usa_tones_11.dat’
FlashHookPeriod = 700
[ControlProtocols Params]
[MGCP Params]
[MEGACO Params]
[SS7 Params]
[Voice Engine Params]
VoiceVolume = 1
MFTransportType = 2
FaxRelayRedundancyDepth = 2
FaxRelayEnhancedRedundancyDepth = 2
BasicRTPPacketInterval = 0
DJBufMinDelay = 70
DJBufOptFactor = 7
RTPRedundancyDepth = 0
RTPPackingFactor = 1
RTPDirectionControl = 0
RFC2833TxPayloadType = 96
RFC2833RxPayloadType = 96
RFC2198PayloadType = 104
FaxBypassPayloadType = 102
EnableStandardSIDPayloadType = 0
AnalogSignalTransportType = 1
[WEB Params]
LogoWidth = ‘339’
[SIP Params]
MAXDIGITS = 3
TIMEBETWEENDIGITS = 5
TIMEFORDIALTONE = 0
ISDTMFUSED = 1
REGISTRATIONTIME = 3600
ISPROXYUSED = 1
AUTHENTICATIONMODE = 1
ISWAITFORDIALTONE = 1
ISTWOSTAGEDIAL = 0
ENABLEHOLD = 1
CDRREPORTLEVEL = 1
MAXACTIVECALLS = 4
GWDEBUGLEVEL = 5
PROXYNAME = '192.168.1.1’
SIPGATEWAYNAME = '192.168.1.125’
USERNAME = 'pstn_in’
CNONCE = '0a123bcf’
PASSWORD = ''
ENABLEVOICEDETECTION = 1
PROGRESSINDICATOR2IP = 1
ALTROUTINGTEL2IPMODE = 0
ISFAXUSED = 1
SUBSCRIPTIONMODE = 1
GWREGISTRATIONNAME = '192.168.1.125’
HOTLINETONEDURATION = 8
ENABLETRANSFER = 1
WAITFORDIALTIME = 400
CODERNAME = g711Ulaw64k,20
PREFIX = 10,192.168.1.1,,0
PREFIX = 20,192.168.1.1,,0
NUMBERMAPIP2TEL = ,0,$$,$$,$$,$$,,$$,192.168.1.1
PSTNPREFIX = ,1,,*,0
TRUNKGROUP_1 = 1-4,4380022,1
PROXYIP = 192.168.1.1
TRUNKGROUPSETTINGS = 1,1
TXDTMFOPTION = 4
ENABLECALLERID_0 = 1
GWLoggerFlags = lgr_flow,D2
GWLoggerFlags = lgr_psbrdif,D2
GWLoggerFlags = lgr_psbrdex,2
GWLoggerFlags = lgr_stack,D2
GWLoggerFlags = lgr_stk_ses,D2
GWLoggerFlags = lgr_stk_mngr,D2
[VXML Params]
[Audio Staging Params]
[PSTN-SDH Params]
– STOP FILE –
Please let me know if I am missing something …
Awaiting your response.
Best regards,
Krish