AudioCodes --
- Thread starter phonebuff
- Start date
drvcrash
Member
- Joined
- Nov 20, 2007
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here is my ini from the audiocodes box and my trunk setup from freepbx
Code:
;**************
;** Ini File **
;**************
;Board: MP-118 FXO
;Serial Number: xxxxxxxxx
;Slot Number: 1
;Software Version: 5.60A.024.003
;DSP Software Version: 204IM => 560.12
;Board IP Address: 192.xxx.xxx.xx
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.xxx.xx.xxx
;Ram size: 32M Flash size: 8M
;Num of DSP Cores: 2 Num DSP Channels: 8
;Profile: NONE
;-----------------------------------------
[SYSTEM Params]
DNSPriServerIP = 208.67.222.222
DNSSecServerIP = 208.67.220.220
TelnetServerEnable = 2
VoiceMenuPassword = 'disable'
[BSP Params]
PCMLawSelect = 3
BaseUDPPort = 10000
LocalMediaDefaultGW = 192.xxx.xx.xxx
LocalMediaIPAddress = 192.xxx.xx.xxx
LocalMediaSubnetMask = 255.255.255.0
LocalControlIPAddress = 192.xxx.xx.xxx
LocalControlSubnetMask = 255.255.255.0
LocalControlDefaultGW = 192.xxx.xx.xxx
LocalOAMIPAddress = 192.xxx.xx.xxx
LocalOAMSubnetMask = 255.255.255.0
LocalOAMDefaultGW = 192.xxx.xx.xxx
StorageServerNetworkAddress = 255.255.255.255
[Analog Params]
[ControlProtocols Params]
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
[Voice Engine Params]
FarEndDisconnectSilenceMethod = 0
CallProgressTonesFilename = 'usa_tones_12.dat'
IdlePCMPattern = 85
DTMFDetectorSensitivity = 1
[WEB Params]
[SIP Params]
ENABLECALLERID = 1
MAXDIGITS = 32
ISPROXYUSED = 1
AUTHENTICATIONMODE = 1
ISWAITFORDIALTONE = 1
ISTWOSTAGEDIAL = 0
ENABLEREVERSALPOLARITY = 1
RADDEBLEVEL = 2
CHANNELSELECTMODE = 1
RADLOGOUTPUT = 1
GWDEBUGLEVEL = 1
ENABLEEARLYMEDIA = 1
DEFAULTNUMBER = ''
PROXYNAME = '192.xxx.xx.xxx'
SIPGATEWAYNAME = '192.xxx.xx.xxx'
USERNAME = 'xxxxxxxxxxxxx'
PASSWORD = 'xxxxxxxxxxxxx'
ENABLEVOICEDETECTION = 1
SUBSCRIPTIONMODE = 1
GWREGISTRATIONNAME = '192.xxx.xx.xxx'
RINGSBEFORECALLERID = 2
WAITFORDIALTIME = 400
DISCONNECTONDIALTONE = 1
[IPsec Params]
[SNMP Params]
;
; *** TABLE DspTemplates ***
; This table contains hidden elements and will not be exposed.
; This table exists on board and will be saved during restarts
;
;
; *** TABLE PREFIX ***
;
;
[ PREFIX ]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID,
PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID;
PREFIX 0 = *, 192.xxx.xxx.xxx, *, 0, 255, 0, -1, , -1, , -1, -1;
[ \PREFIX ]
;
; *** TABLE CoderName ***
;
;
[ CoderName ]
FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval, CoderName_rate, CoderName_PayloadType, CoderName_Sce;
CoderName 0 = g711Ulaw64k, 20, 0, 255, 0;
[ \CoderName ]
;
; *** TABLE TrunkGroup ***
;
;
[ TrunkGroup ]
FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber,
TrunkGroup_ProfileId, TrunkGroup_LastTrunkId, TrunkGroup_Module;
TrunkGroup 0 = 0, 255, 1, 4, 410-xxx-xxxx, 0, 255, 255;
TrunkGroup 1 = 0, 255, 5, 8, 410-xxx-xyxx, 0, 255, 255;
[ \TrunkGroup ]
;
; *** TABLE ProxyIp ***
;
;
[ ProxyIp ]
FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId;
ProxyIp 0 = 192.xxx.xx.xxx, 0, 0;
[ \ProxyIp ]
;
; *** TABLE TxDtmfOption ***
;
;
[ TxDtmfOption ]
FORMAT TxDtmfOption_Index = TxDtmfOption_Type;
TxDtmfOption 0 = 4;
[ \TxDtmfOption ]
;
; *** TABLE TrunkGroupSettings ***
;
;
[ TrunkGroupSettings ]
FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode,
TrunkGroupSettings_GatewayName, TrunkGroupSettings_ContactUser, TrunkGroupSettings_ServingIPGroup;
TrunkGroupSettings 0 = 1, 1, 255, , , -1;
[ \TrunkGroupSettings ]
;
; *** TABLE EnableCallerId ***
;
;
[ EnableCallerId ]
FORMAT EnableCallerId_Index = EnableCallerId_IsEnabled;
EnableCallerId 0 = 1;
EnableCallerId 1 = 1;
EnableCallerId 2 = 1;
EnableCallerId 3 = 1;
EnableCallerId 4 = 1;
EnableCallerId 5 = 1;
EnableCallerId 6 = 1;
EnableCallerId 7 = 1;
[ \EnableCallerId ]
;
; *** TABLE TargetOfChannel ***
;
;
[ TargetOfChannel ]
FORMAT TargetOfChannel_Index = TargetOfChannel_Destination, TargetOfChannel_Type;
TargetOfChannel 0 = 410xxxxxxx, 1;
TargetOfChannel 1 = 410xxxxxxx, 1;
TargetOfChannel 2 = 410xxxxxxx, 1;
TargetOfChannel 3 = 410xxxxxxx, 1;
TargetOfChannel 4 = 410xxxxyxx, 1;
TargetOfChannel 5 = 410xxxxyxx, 1;
TargetOfChannel 6 = 410xxxxyxx, 1;
TargetOfChannel 7 = 410xxxxyxx, 1;
[ \TargetOfChannel ]
;
; *** TABLE ProxySet ***
;
;
[ ProxySet ]
FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap;
ProxySet 0 = 0, 60, 0, 0;
[ \ProxySet ]
Okay I made the changes you suggested and updated my definitions. But no luck. On the MP-118 I do see activity it rings three times and then a quick answer / hangup..No Asterisk activity that I can see even with SIP debug on .
PBX IN A Flash = 1.7.5.6
FreePBX = 2.8.1.4
Asterisk = 1.8.4.1
Log is Activated
8d:23h:27m:54s ( lgr_psbrdex)(47 ) recv <-- ANALOG_IF_RING_START Ch:7 type(0)
8d:23h:27m:56s ( lgr_psbrdex)(48 ) recv <-- EV_ANALOG_IF_RING_END Ch:7 type(0)
8d:23h:27m:57s ( lgr_psbrdex)(49 ) recv <-- EV_DETECT_CALLER_ID Ch:7 (Name=COMPUDESIGNS , Number=)
8d:23h:28m:0s ( lgr_psbrdex)(50 ) recv <-- ANALOG_IF_RING_START Ch:7 type(0)
8d:23h:28m:2s ( lgr_psbrdex)(51 ) recv <-- EV_ANALOG_IF_RING_END Ch:7 type(0)
8d:23h:28m:6s ( lgr_psbrdex)(52 ) recv <-- ANALOG_IF_RING_START Ch:7 type(0)
8d:23h:28m:8s ( lgr_psbrdex)(53 ) recv <-- EV_ANALOG_IF_RING_END Ch:7 type(0)
8d:23h:28m:8s ( lgr_psbrdif)(54 ) ActivateDigitMap for channel : 7, MaxDialStringLength = 32, MaxEndDialTimer = 4000,
MaxLongInterDigitTimer = 8000, MaxStartTimer = 16000, DigitMap = [0-9*#ABCD][0-9ABCD].T, DialPlanIndex = -1
8d:23h:28m:8s ( lgr_psbrdif)(55 ) UpdateChannelParams, Channel 7
8d:23h:28m:8s ( lgr_psbrdif)(56 ) Turn ringer ON for channel 7
8d:23h:28m:8s ( lgr_psbrdex)(57 ) PCIIFChangeChannelParams failed ECNlpMode
8d:23h:28m:8s ( lgr_psbrdif)(58 ) Changed ECNlpMOde to: 1
8d:23h:28m:10s ( lgr_flow)(59 ) ---- Incoming SIP Message from 10.179.131.160:5060 to SIPInterface #0 ----
8d:23h:28m:10s OPTIONS sip:10.179.131.161 SIP/2.0
Via: SIP/2.0/UDP 10.179.131.160:5060;branch=z9hG4bK296162b0
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as59d4e64a
To: <sip:10.179.131.161>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.4.1)
Date: Wed, 20 Jul 2011 23:25:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
8d:23h:28m:10s ( lgr_psbrdif)(61 ) QueryOnHookPortStatus (ChannelNum=0), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(62 ) QueryOnHookPortStatus (ChannelNum=1), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(63 ) QueryOnHookPortStatus (ChannelNum=2), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(64 ) QueryOnHookPortStatus (ChannelNum=3), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(65 ) QueryOnHookPortStatus (ChannelNum=4), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(66 ) QueryOnHookPortStatus (ChannelNum=5), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(67 ) QueryOnHookPortStatus (ChannelNum=6), status = 0 Polarity = 0
8d:23h:28m:10s ( lgr_psbrdif)(68 ) QueryOnHookPortStatus (ChannelNum=7), status = 1 Polarity = 0
8d:23h:28m:10s ( lgr_flow)(69 ) ---- Outgoing SIP Message to 10.179.131.160:5060 from SIPInterface #0 ----
8d:23h:28m:10s SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.179.131.160:5060;branch=z9hG4bK296162b0
From: "Unknown" <sip:[email protected]>;tag=as59d4e64a
To: <sip:10.179.131.161>;tag=1c1312966016
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected]:5060>
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
X-Resources: telchs=0/1;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 234
v=0
o=AudiocodesGW 1312969648 1312969523 IN IP4 10.179.131.161
s=Phone-Call
c=IN IP4 10.179.131.161
t=0 0
m=audio 6000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
8d:23h:28m:19s ( lgr_psbrdex)(71 ) recv <-- CPTONE_DETECT Ch:7 Ind:0 Type:1 DIAL_TONE
8d:23h:28m:24s ( lgr_psbrdex)(72 ) recv <-- EV_DIALED_STRING Ch:7 Str: MapNum:-2 CMM Match:0 EI:
8d:23h:28m:24s ( lgr_psbrdex)(73 ) InsertBoardEvent- event 107 inserted channel 7
8d:23h:28m:24s ( lgr_psbrdif)(74 ) #7:cpDigitMapHndlr_Stop - Stoped (0)
8d:23h:28m:24s ( lgr_psbrdif)(75 ) RFC2833RTPPayloadType: Rx=96 Tx=96 DTMF Transport=3
8d:23h:28m:24s ( lgr_psbrdif)(76 ) #7:Channel will be open WITH DSP
8d:23h:28m:24s ( lgr_psbrdif)(77 ) Turn ringer OFF for channel 7
8d:23h:28m:25s ( lgr_psbrdex)(78 ) InsertBoardEvent- event 116 inserted channel 7
8d:23h:28m:25s ( lgr_psbrdif)(79 ) #7:cpDigitMapHndlr_Stop - Stoped (0)
8d:23h:28m:25s ( lgr_psbrdif)(80 ) RFC2833RTPPayloadType: Rx=96 Tx=96 DTMF Transport=3
8d:23h:28m:25s ( lgr_psbrdif)(81 ) #7:Channel will be open WITH DSP
sip_additional.conf
[MP-118]
disallow=all
type=friend
allow=ulaw
context=from-pstn
username=axxxxxxx
secret=xxxxxxxx
dtmfmode=inband
host=10.179.131.161
nat=no
canreinvite=no
qualify=yes
drvcrash
Member
- Joined
- Nov 20, 2007
- Messages
- 108
- Reaction score
- 0
another problem i had in the beginning also was failtoban kept banning the audiocodes box. Check and see if it shows up in sip peers.
there is also a setting under endpoint settings called Automatic dialing and i have each channel set to enabled with its trunk telephone number as the destination number
there is also a setting under endpoint settings called Automatic dialing and i have each channel set to enabled with its trunk telephone number as the destination number
Update --
With some very much appreciated help from DRVCRASH the MP-118 is passing calls to Asterisk. (Have not tried outbound yet) When I have a second I will scrub the "ini" backup file from the working unit and post it here.
I am also in touch with the company and may attempt a white paper how-to and You Tube to go with the WP.
Again special thanks to DRVCRASH.
------
With some very much appreciated help from DRVCRASH the MP-118 is passing calls to Asterisk. (Have not tried outbound yet) When I have a second I will scrub the "ini" backup file from the working unit and post it here.
I am also in touch with the company and may attempt a white paper how-to and You Tube to go with the WP.
Again special thanks to DRVCRASH.
------
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