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Audiocodes MP118 Caller ID


Worm78
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I have started using an MP118 as a trunk on its own domain. This was used as a backup trunk using the dialplan feature. I added this to its own domain and all works ok for incoming and outgoing calls except CLID. Everything shows up as 1000 or 1001. I have hooked a caller id enabled phone directly to the pots line to ensure the telco is sending info and this works.

 

I see nothing in the logs on the MP118 when calling in that has a phone number.

 

Under end point settings

 

I'm using the automatic dial feature to send it to the trunk ACD which is 700. Caller ID is set to enabled on each fxo. Detect clid from telco is as well enabled

 

 

End point phone numbers is blank. I'm using three centrex lines and not rolling or using hunt groups on the MP.

Device is setup in proxy mode.

 

Any ideas?

 

 

Thanks,

Brian

 

 

 

 

 

 

here is the ini file and unit is on latest firmware. PBX Version: 3.4.0.3201 (Win32)

 

 

;**************

;** Ini File **

;**************

 

;Board: MP-118 FXO

;Serial Number: 762960

;Slot Number: 1

;Software Version: 5.00A.024

;Board IP Address: 192.168.1.253

;Board Subnet Mask: 255.255.255.0

;Board Default Gateway: 192.168.1.1

;Ram size: 32M Flash size: 8M

;Num DSPs: 2 Num DSP channels: 8

;Profile: NONE

;------------------------------

 

 

[sYSTEM Params]

 

SyslogServerIP = 10.1.1.89

VXMLFIleName = ''

VoiceMenuPassword = 'disable'

 

[bSP Params]

 

PCMLawSelect = 3

LocalOAMIPAddress = 192.168.1.253

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

VoiceVolume = 3

InputGain = 3

DTMFVolume = 0

RFC2833PayloadType = 101

 

[WEB Params]

 

LogoWidth = '339'

 

[sIP Params]

 

ENABLECALLERID = 1

MAXDIGITS = 11

LOCALSIPPORT = 5060

PLAYRBTONE2IP = 0

REGISTRATIONTIME = 3600

SIPT1RTX = 500

SIPT2RTX = 4000

ISPROXYUSED = 1

SIPDESTINATIONPORT = 5060

PLAYRBTONE2TEL = 2

ISTWOSTAGEDIAL = 0

DETFAXONANSWERTONE = 0

ENABLECURRENTDISCONNECT = 1

CHANNELSELECTMODE = 1

GWDEBUGLEVEL = 5

ENABLERPIHEADER = 1

ENABLEEARLYMEDIA = 1

ISUSERPHONE = 0

SIPSESSIONEXPIRES = 0

SIPGATEWAYNAME = '192.168.1.253'

CNONCE = '0a123bcf'

PASSWORD = '787899'

PRACKMODE = 1

SIPMAXRTX = 7

ASSERTEDIDMODE = 0

ISUSERPHONEINFROM = 0

ADDTON2RPI = 1

USESOURCENUMBERASDISPLAYNAME = 1

MINSE = 90

IPALERTTIMEOUT = 180

ISFAXUSED = 1

SIPTRANSPORTTYPE = 0

TCPLOCALSIPPORT = 5060

RINGSBEFORECALLERID = 2

TLSLOCALSIPPORT = 5061

ENABLESIPS = 0

USERAGENTDISPLAYINFO = ''

SESSIONEXPIRESMETHOD = 0

USEDISPLAYNAMEASSOURCENUMBER = 0

USETELURIFORASSERTEDID = 0

USESIPTGRP = 0

SIPSUBJECT = ''

CODERNAME = g711Ulaw64k,20,0,$$,0

PREFIX = *,192.168.1.254,*,0,255

TARGETOFCHANNEL0 = 700,1

TARGETOFCHANNEL1 = 700,1

TARGETOFCHANNEL2 = 700,1

TARGETOFCHANNEL3 = 700,1

TARGETOFCHANNEL4 = 700,1

TARGETOFCHANNEL5 = 700,1

TARGETOFCHANNEL6 = 700,1

TARGETOFCHANNEL7 = 700,1

TRUNKGROUP = 1-1,,0

TRUNKGROUP = 2-2,,0

TRUNKGROUP = 3-3,,0

TRUNKGROUP = 4-4,,0

TRUNKGROUP = 5-5,,0

PROXYIP = 192.168.1.254

TXDTMFOPTION = 4

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]

 

 

[PSTN-SDH Params]

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Number dialed in form was 985 8299 and .253 is the Audiocodes, 700 is a hunt group.

 

 

While I'm asking I'm also having an issue where intermittently I get no audio or one way audio (outgoig only) when the system cfwd's to a cell phone. No other issues with the trunk. It seems to happen 1 out of 4 times. I have also had a very garbled sound a few times.

 

 

[5] 2009/08/17 11:31:53: Identify trunk (IP address and DID match) 1

[7] 2009/08/17 11:31:53: Set packet length to 20

[6] 2009/08/17 11:31:53: Sending RTP for 66993680812102000194319@192.168.1.253#a7b0f0dfd1 to 192.168.1.253:6000

[5] 2009/08/17 11:31:53: Trunk Audiocodes (not global) sends call to account 700 in domain realty

[7] 2009/08/17 11:31:53: Looking for EPID 700

[7] 2009/08/17 11:31:53: Set packet length to 20

[6] 2009/08/17 11:31:53: Send codec pcmu/8000

[7] 2009/08/17 11:31:53: Call a99082be@pbx#16196: Clear last request

[7] 2009/08/17 11:31:55: Call a99082be@pbx#16196: Clear last INVITE

[6] 2009/08/17 11:31:55: Send codec=pcmu/8000 afrer answer

[6] 2009/08/17 11:31:55: Sending RTP for a99082be@pbx#16196 to 192.168.1.9:61370

[7] 2009/08/17 11:31:55: Determine pass-through mode after receiving response

[7] 2009/08/17 11:31:55: a99082be@pbx#16196: RTP pass-through mode

[7] 2009/08/17 11:31:55: 66993680812102000194319@192.168.1.253#a7b0f0dfd1: RTP pass-through mode

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Could you please enable the "Log Other Messages" under the SIP logging and send the log to support@pbxnsip.com?

 

Number dialed in form was 985 8299 and .253 is the Audiocodes, 700 is a hunt group.

 

 

While I'm asking I'm also having an issue where intermittently I get no audio or one way audio (outgoig only) when the system cfwd's to a cell phone. No other issues with the trunk. It seems to happen 1 out of 4 times. I have also had a very garbled sound a few times.

 

 

[5] 2009/08/17 11:31:53: Identify trunk (IP address and DID match) 1

[7] 2009/08/17 11:31:53: Set packet length to 20

[6] 2009/08/17 11:31:53: Sending RTP for 66993680812102000194319@192.168.1.253#a7b0f0dfd1 to 192.168.1.253:6000

[5] 2009/08/17 11:31:53: Trunk Audiocodes (not global) sends call to account 700 in domain realty

[7] 2009/08/17 11:31:53: Looking for EPID 700

[7] 2009/08/17 11:31:53: Set packet length to 20

[6] 2009/08/17 11:31:53: Send codec pcmu/8000

[7] 2009/08/17 11:31:53: Call a99082be@pbx#16196: Clear last request

[7] 2009/08/17 11:31:55: Call a99082be@pbx#16196: Clear last INVITE

[6] 2009/08/17 11:31:55: Send codec=pcmu/8000 afrer answer

[6] 2009/08/17 11:31:55: Sending RTP for a99082be@pbx#16196 to 192.168.1.9:61370

[7] 2009/08/17 11:31:55: Determine pass-through mode after receiving response

[7] 2009/08/17 11:31:55: a99082be@pbx#16196: RTP pass-through mode

[7] 2009/08/17 11:31:55: 66993680812102000194319@192.168.1.253#a7b0f0dfd1: RTP pass-through mode

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Could you please enable the "Log Other Messages" under the SIP logging and send the log to support@pbxnsip.com?

 

Based on the log you have sent, there is no-way the phone would display either 1000 or 1001.

You should see ""INDIANA CALL " 3179858299" as the caller(of course, you may see 700 or 700 (3179858299) depending on the "From Header" setting on the ACD)

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I was told it said that once as well. I have not seen it. I do see it on the log I sent which is really weird. I just tried it and I get 1000.

 

Trunk rolls to a hunt group (700) for 6 seconds which rings the receptionist extension 101, and then goes to the ACD which is 701

 

 

I checked the From-Header: on the hunt group and it is set to calling party

I didn't see an option on the ACD

 

I will resend the log from 5 minutes ago where I just saw the 1000 come in.

 

 

EDIT ******** Just founf if I call the other lines coming in it works. Calling the main line it does not. May be a telephone company issue. Grrrrrrrrrrrrrrrr *********

 

 

 

 

 

 

 

 

 

Here is the call log as well

 

 

2009/08/19 09:49:13 1000 (1000@192.168.1.253) 700 00:35

2009/08/19 09:50:15 1000 John Doe (303)

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Ok. Got the new log. Based on that, I can understand why you would see '1000' as the caller. The INVITE from Audiocodes to pbx has below as the 'from' header.

 

From: "1000" <sip:1000@192.168.1.253>;tag=1c1759636984

 

If the phone company is delivering the caller id, then AC should pass it to the PBX (may be you can connect a phone to the line and see whether the caller id shows up).

 

 

 

I was told it said that once as well. I have not seen it. I do see it on the log I sent which is really weird. I just tried it and I get 1000.

 

Trunk rolls to a hunt group (700) for 6 seconds which rings the receptionist extension 101, and then goes to the ACD which is 701

 

 

I checked the From-Header: on the hunt group and it is set to calling party

I didn't see an option on the ACD

 

I will resend the log from 5 minutes ago where I just saw the 1000 come in.

 

 

EDIT ******** Just founf if I call the other lines coming in it works. Calling the main line it does not. May be a telephone company issue. Grrrrrrrrrrrrrrrr *********

 

 

 

 

 

 

 

 

 

Here is the call log as well

 

 

2009/08/19 09:49:13 1000 (1000@192.168.1.253) 700 00:35

2009/08/19 09:50:15 1000 John Doe (303)

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