mattlandis
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Posts posted by mattlandis
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Yea, we removed it. If you run version 3.2, you can just put "file:cdr.txt" into the SOAP CDR URL field and you'll get nicely formatted files in the file system.
THanks for that reply.
Is it possible to put it on a NAS/or another server drive?
That way we can pull it into a SQL database and do our KPI's on it from there.
so is the cdr tool going away?
thanks,
matt
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It seems all the links to the cdr tool are dead. the wiki link, and links in this forum.
Or am I being blind? ;-)
What I'm looking for is a simple way to see certain info: calls per day; call per ext. etc.
matt
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Hi,
I have it set to press 1 but it doesn't.
I removed cell# and saved and added it again to make sure nothing is unhitched but it still doesnt ask to press 1.
I know it did before but it stopped sometime.
(senario: from office someone presses *00<ext>)
tx
matt
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Hi,
Just a beta report:
i can login but get error trying to monitor exts.
Followed wiki.
DND seems to work.
matt
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hello,
I found the solution:
http://forum.pbxnsip.com/index.php?s=&...post&p=8329
in the auto attendent > direct destinations > user input
you need to add a # after the 1 to make it 1#.
(the # delays so it can accept an extension such as 101, 102, etc)
thanks
matt
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Hi,
I have 2 trunks: first one is is a 4line pstn gateway and 2nd one is a voipline we have 4 co lines assigned to it.
our dial plan is simple:
pstngateway, *,
voipline, *,
When all the lines on pstngateway (4 lines) are full, and we dial out we get a busy signal instead of it rolling over to the voipline.
What am I doing wrong?
Thanks,
matt
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I have exactly the same problem as above but I don't understand the resolution.
I have extensions from 101-150. My AA says press 1 for sales and 2 for service...
I have extension input=after 3 digit...
when someone calls in and presses 101 to go directly to ext. 101 they go to salesman...who is being inundated...
What specifically do you suggest as a work around?
Thanks,
matt
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Selling Professionals.
If you are out their selling products you are going to be forced to compare apples and oranges. Instead, our clients have never seen the name PBXnSIP unless they actually pay attention when we are training them on the WEB login page for their extension.
In the last 25 years of IT sales/support, whenever the client controls the debate of 1 vs. 2, I choose to walk away from the opportunity. Unless the customer is openly willing to place value on a solution to a business problem, then we really have nothing to offer. Getting a client to bleed with pain about their troubles, is the only way to become the Doctor that puts them whole again. Stop selling products and start selling solutions. If you can charge a fair and going rate to support a free product for a client, and it's meets their needs and you sleep at night - Go for it. There's nothing wrong with making a profit. However, after funding 1.5 years worth of Asterisk development and a large install that consumed 300+ hours over the first year, I choose to move all PBX sales to PBXnSIP. Our Asterisk developer was tightly involved and did some great open source work on my dime. The client never new it was Asterisk, and when we switched to PBXnSIP they never missed a beat and our support hours dramatically fell.
Very interesting comments. Thanks.
matt
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Those files are in the audio_moh directory. Check "bi_gong1.wav" and "bi_gong2.wav".
Just feedback: I think the activated and deactivated sound file sounds very professional.
matt
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I have nothing after that, if you post your audiocodes and PBXnSIP logs, I am sure we can figure out your problem.
I've worked with pbxnsip on this and they have provided a solution. So I want to say a public "thanks" to them.
I will post my .ini too.
I appreciate your willingness to help.
matt
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The IP of the pbxnsip is the 192.168.0.10, I don't know what that password is, it certainly is is not to the PBXnsip, as it is a gateway trunk not registered.
According to http://wiki.pbxnsip.com/index.php/Audiocodes those 2 values should be left at their default values.
Hi,
Thanks for sending that mp114 ini file.
Unfortunately it didn't work.
My ini has quite a bit of info after the "[ \TargetOfChannel ]". there's no chance the forum cut off the bottom of the file? If so your welcome to email me directly.
matt
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Hello,
Thanks for that post...that is what I was looking for...
I'm not at the computer to test it but had some questions:
#1- CNONCE = '0a123bcf' and PASSWORD = '787899'. Is this a password setup in pbxnsip?
#2- is the ip address of the pbxnsip server in this senario 192.168.0.10?
My config is here:
http://forum.pbxnsip.com/index.php?showtopic=2003
if you want to critique it.
Once again, thanks.
matt
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Yes, the Blockview is missing....
The Listview is all twitchy and flashy
It shimmys and shakes like the San Andreas Fault.
What was lacking in the Blockview??? I liked it..
Bill H
Very interesting (graphic) explanation...chuckle...
matt
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The audiocodes will be MP-11X, you can follow the Wiki to configure it.
hi,
I have an audiocodes mp-114 fxo and i can't seem to get it working with the wiki config.
(I had it working on another system)
i've posted the .ini if you want to see if there are obvious mistakes on it.
tx
matt
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We have good experience with AudioCodes, Mediatrix, Parlay, Patton, Vegastream as seperate gateways. Also there is the AVM FritzBox if you want BRI. If you want to use PC cards, there is Sangoma and Dialogic that offer SIP-based cards. We try to keep http://wiki.pbxnsip.com/index.php/Interoperability_Pages up to date.
Thanks for that reply.
I'm looking on your interop page and it seems like things are a bit out of date:
-i can't find the audiocodes MP-10x mentioned on the wiki on audiocodes.com--it seems like a discontinued model; the wiki is posted 2006
-when i click on the "Sangoma" link I get a "login required to edit" page
Perhaps i'm looking at the wrong place?
thanks for the help,
matt
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multi domain - it is not registrered any where.....
Hi,
Does this happen often? In other words is this something to watch out for?
tx
matt
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I have removed a trunk in a domain on the PBXnSIP, BUT now, 14 days later, it STILL try to reg. to the SIP server.
Trunk 29 (11111111) has outbound proxy udp:XX.XX.XXX.XXX:5060
failed to reg.
Any ideas what could be wrong?
hello,
What happens if you restart the pbxnsip service?
matt
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Hello,
Here is how I have my audiocodes MP-114 setup but it doesn't dial out or in.
My PBxnsip server=192.168.1.152.
My audiocodes mp114=192.168.1.151.
My trunk in pbxnsip equals the default setup with below changed:
domain=192.168.1.151
and username and password= blank.
Default dial rule.
thanks!
matt
;**************
;** Ini File **
;**************
;Board: MP-114 FXO
;Serial Number: 905231
;Slot Number: 1
;Software Version: 5.20A.031.007
;DSP Software Version: 204IM => 520.13
;Board IP Address: 192.168.1.151
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.168.1.1
;Ram size: 32M Flash size: 8M
;Num DSPs: 1 Num DSP channels: 4
;Profile: NONE
;-----------------------------------------
[sYSTEM Params]
SyslogServerIP = 192.168.1.151
EnableSyslog = 1
[bSP Params]
PCMLawSelect = 3
LocalOAMIPAddress = 192.168.1.151
LocalOAMSubnetMask = 255.255.255.0
LocalOAMDefaultGW = 192.168.1.1
[ATM Params]
[Analog Params]
FXSLoopCharacteristicsFilename = 'MP11x-02-1-FXS_16KHZ.dat'
[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
[PSTN Params]
[sS7 Params]
[Voice Engine Params]
IdlePCMPattern = 85
CallerIDType = 1
V21ModemTransportType = 1
V22ModemTransportType = 1
V23ModemTransportType = 1
V32ModemTransportType = 1
V34ModemTransportType = 1
CallProgressTonesFilename = 'usa_tones_12.dat'
[WEB Params]
LogoWidth = '339'
[sIP Params]
MAXDIGITS = 49
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
ISTWOSTAGEDIAL = 0
DETFAXONANSWERTONE = 1
ENABLECURRENTDISCONNECT = 1
CHANNELSELECTMODE = 2
PROXYNAME = '192.168.1.152'
REGISTRARIP = '192.168.1.152'
SIPGATEWAYNAME = 'MP114-4FXO'
ENABLEVOICEDETECTION = 1
PRACKMODE = 0
ADDTON2RPI = 0
ISFAXUSED = 1
HOTLINETONEDURATION = 0
[VXML Params]
[iPsec Params]
[Audio Staging Params]
ChassisPhysicalAlias = ''
ChassisPhysicalAssetID = ''
ifAlias_0 = ''
ifAlias_1 = ''
ifAlias_2 = ''
ifAlias_3 = ''
ifAlias_4 = ''
ifAlias_5 = ''
ifAlias_6 = ''
ifAlias_7 = ''
ifAlias_8 = ''
ifAlias_9 = ''
;
; *** TABLE DspTemplates ***
; This table contains hidden elements and will not be exposed.
; This table exists on board and will be saved during restarts
;
;
; *** TABLE CoderName ***
;
;
[ CoderName ]
FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval, CoderName_rate, CoderName_PayloadType, CoderName_Sce;
CoderName 0 = g711Ulaw64k, 20, 0, 0, 0;
CoderName 1 = g711Alaw64k, 20, 10, 8, 0;
CoderName 2 = t38fax, 255, 255, 0, 255;
CoderName 3 = g729, 20, 0, 18, 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, 1, 10001, 1, 255, 255;
TrunkGroup 1 = 0, 255, 2, 2, 10002, 1, 255, 255;
TrunkGroup 2 = 0, 255, 3, 3, 10003, 1, 255, 255;
TrunkGroup 3 = 0, 255, 4, 4, 10004, 1, 255, 255;
[ \TrunkGroup ]
;
; *** TABLE ProxyIp ***
;
;
[ ProxyIp ]
FORMAT ProxyIp_Index = ProxyIp_IpAddress;
ProxyIp 0 = 192.168.1.152;
[ \ProxyIp ]
;
; *** TABLE TxDtmfOption ***
;
;
[ TxDtmfOption ]
FORMAT TxDtmfOption_Index = TxDtmfOption_Type;
TxDtmfOption 0 = 4;
[ \TxDtmfOption ]
;
; *** TABLE TelProfile ***
;
;
[ TelProfile ]
FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone;
TelProfile 1 = "Default Tel Profile", 1, 0, 1, 10, 10, 46, 46, 0, 0, 0, 0, 1, 0, 1, 0, 0, 400, 0, 0, 255;
[ \TelProfile ]
;
; *** TABLE IpProfile ***
;
;
[ IpProfile ]
FORMAT IpProfile_Index = IpProfile_ProfileName, IpProfile_IpPreference, IpProfile_CodersGroupID, IpProfile_IsFaxUsed, IpProfile_JitterBufMinDelay, IpProfile_JitterBufOptFactor, IpProfile_IPDiffServ, IpProfile_SigIPDiffServ, IpProfile_SCE, IpProfile_RTPRedundancyDepth, IpProfile_RemoteBaseUDPPort, IpProfile_CNGmode, IpProfile_VxxTransportType, IpProfile_NSEMode, IpProfile_IsDTMFUsed, IpProfile_PlayRBTone2IP, IpProfile_EnableEarlyMedia, IpProfile_ProgressIndicator2IP, IpProfile_EnableEchoCanceller, IpProfile_CopyDest2RedirectNumber;
IpProfile 1 = , 1, 0, 1, 10, 10, 46, 40, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1, 1, 0;
[ \IpProfile ]
;
; *** TABLE EnableCallerId ***
;
;
[ EnableCallerId ]
FORMAT EnableCallerId_Index = EnableCallerId_IsEnabled;
EnableCallerId 0 = 1;
EnableCallerId 1 = 1;
EnableCallerId 2 = 1;
EnableCallerId 3 = 1;
[ \EnableCallerId ]
;
; *** TABLE CallerDisplayInfo ***
;
;
[ CallerDisplayInfo ]
FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, CallerDisplayInfo_IsCidRestricted;
CallerDisplayInfo 0 = 10001, 0;
CallerDisplayInfo 1 = 10002, 0;
CallerDisplayInfo 2 = 10003, 0;
CallerDisplayInfo 3 = 10004, 0;
[ \CallerDisplayInfo ]
;
; *** TABLE TargetOfChannel ***
;
;
[ TargetOfChannel ]
FORMAT TargetOfChannel_Index = TargetOfChannel_Destination, TargetOfChannel_Type;
TargetOfChannel 0 = 10001, 1;
TargetOfChannel 1 = 10002, 1;
TargetOfChannel 2 = 10003, 1;
TargetOfChannel 3 = 10004, 1;
[ \TargetOfChannel ]
;
; *** TABLE Authentication ***
;
;
[ Authentication ]
FORMAT Authentication_Index = Authentication_UserId, Authentication_UserPassword;
Authentication 0 = 10001, 10001;
Authentication 1 = 10002, 10002;
Authentication 2 = 10003, 10003;
Authentication 3 = 10004, 10004;
[ \Authentication ]
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Is there a recommended list somewhere?
tx
matt
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I saw the wiki pages but they seem very old and not very specific/clear.
i'm talking about http://wiki.pbxnsip.com/index.php/Audiocodes
Is there something I'm missing?
tx
matt
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hi,
has anyone gotten the xlite presence to consistently work?
tx
matt
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Hi,
I would agree with this suggestion.
matt
I don't get "press 1 to answer" on my cell phone
in Extension Setup
Posted
I was thinking of the senario that a collegue transfers a call to *00<ext> when a person is on the road...
so the only time it works is if the cellphone is included in rining when the extension does?
that fine, didn't realize that.
tx
matt