mattlandis

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

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

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

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

Hi, Just a beta report: i can login but get error trying to monitor exts. Followed wiki. DND seems to work. matt

http://wiki.pbxnsip.com/index.php/PAC_V3 matt

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

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

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

Very interesting comments. Thanks. matt

Just feedback: I think the activated and deactivated sound file sounds very professional. matt

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

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

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

Very interesting (graphic) explanation...chuckle... matt

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

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

Hi, Does this happen often? In other words is this something to watch out for? tx matt

hello, What happens if you restart the pbxnsip service? matt

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 ]

Is there a recommended list somewhere? tx matt

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

hi, has anyone gotten the xlite presence to consistently work? tx matt

Hi, I would agree with this suggestion. matt