Hi, I’m kind a new to audiocodes interoperating with asterisk. I having problems regarding the asterisk connecting to audiocodes to a PABX. Our Pabx system can call to PSTN by dialing 9 then a dial tone will be heard and dial the 7 digit number. I have connected the audiocodes to the PABX system a place a fxs phone. I dialed the same as with the PABX and It was successful. the structure would be this. IP Phone with FXS -->audiocodes–>PABX–>POTS. By when I have connected the asteisk with audiocodes, the structure would be like this,
hosted softhone–>asterisk–>audiocodes–>PABX–>POTS. When I dialed 9 i did not get any dial tone which I think a requirement for the PABX. I cannot make a call when I dialed 9xxxxxxx. The calls from my softhone which is hosted by asterisk passes the call to asterisk, then asterisk passes calls to quintum. Then quintum passes calls to the PABX but returns an error. Unallocated number error. How can I have a dial tone by pressing 9 in asterisk. Please help. Below is my asterisk configuration.
sip.conf
[audiocodes]
context=audioc-out
type=friend
username=audiocodes
secret=audiocodes
host=192.168.100.66
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
nat=yes
dtmfmode=rfc2833
[5303]
context=audio-sip
type=friend
username=5303
secret=5303
host=dynamic
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
nat=yes
extensions.conf
[audioc-out]
exten=>_9X.,1,Dial(SIP/${EXTEN}@192.168.100.66,50,ro)
exten=>_9X.,n,Hangup
include=>audio-sip
[audio-sip]
exten=>5303,1,Dial(SIP/5303,50,ro)
exten=>5303,n,Hangup
include=>audioc-out
Here is the config file for the audiocodes mediant 1000
;**************
;** Ini File **
;**************
;Board: Mediant 1000
;Serial Number: 675975
;Slot Number: 1
;Software Version: 4.80A.027.002
;Board IP Address: 192.168.100.66
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.168.100.254
;Ram size: 128M Flash size: 32M
;Num DSPs: 8 Num DSP channels: 48
;Profile: NONE
;------------------------------
[SYSTEM Params]
DNSPriServerIP = 10.2.1.2
SyslogServerIP =
EnableSyslog = 1
DisableRS232 = 1
[BSP Params]
PCMLawSelect = 1
TDMBusClockSource = 4
LocalOAMIPAddress = 192.168.100.66
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 = ‘M2k_usa_tones.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
[PSTN Params]
TraceLevel = 1
ProtocolType = 7
FramingMethod = c
LineCode = 2
CASTablesNum = 5
CASFileName_0 = 'R2_Philippines_new_ANI.dat’
CASFileName_1 = 'R2_Philippines_new_NoANI.dat’
CASFileName_2 = 'R2_Philippines_NoANI.dat’
CASFileName_3 = 'R2_Philippines_ANI.dat’
CASFileName_4 = 'R2_Philippines02_ANI.dat’
CASFileName_5 = ''
CASFileName_6 = ''
CASFileName_7 = ‘’
[SS7 Params]
[Voice Engine Params]
IdlePCMPattern = 255
VoiceVolume = 1
FaxRelayRedundancyDepth = 2
FaxRelayEnhancedRedundancyDepth = 2
RFC2833PayloadType = 101
[WEB Params]
LogoWidth = ‘339’
[SIP Params]
TIMEFORREORDERTONE = 5
REGISTRATIONTIME = 3600
ISPROXYUSED = 1
CDRREPORTLEVEL = 2
GWDEBUGLEVEL = 5
ENABLEEARLYMEDIA = 1
PROXYNAME = 'imxccm’
REGISTRARIP = '192.168.100.5’
SIPGATEWAYNAME = 'audiocodes’
USERNAME = 'audiocodes’
CNONCE = '0a123bcf’
PASSWORD = 'audiocodes’
STATICNATIP =
ISFAXUSED = 1
GWREGISTRATIONNAME = 'audiocodes’
REGISTRARNAME = 'audiocodes’
CODERNAME = g711Alaw64k,20,0,$$,0
CODERNAME = g711Ulaw64k,20,0,$$,0
PREFIX = ,192.168.100.5,,0,255
NUMBERMAPTEL2IP = 5103,4,4,$$,$$,$$,36035103,$$,*
PSTNPREFIX = ,1,,*,0
TRUNKGROUP_1 = 0-0/1-10,1100,0
TRUNKGROUP_1 = 1-31,1200,0
TRUNKGROUP_1 = 1-31,1300,0
TRUNKGROUP_1 = 1-31,1400,0
PROXYIP = 192.168.100.5
TRUNKGROUPSETTINGS = 1,1
TXDTMFOPTION = 4
[SCTP Params]
[VXML Params]
[IPsec Params]
[Audio Staging Params]
[PSTN-SDH Params]
Please help me. Thank you very much
Best Regards,
newbie_aste