Friedom-Tech

please see attached wireshark trace....im trying to attach it!!!!doesnt allow me to!!!

im looking in the log now and for some reason now i see that it is playing rinback.wav...could this be an issue with my sangoma card?

im sorry but im not understanding what you are saying. the log level on both pbx's are set with exactly the same settings and when i look in the list of active calls i see the incoming call and once the person picks up audio both ways is fine.... please advise exactly what you want me to do to further troubleshoot this.

i had those folders in there but redownloaded them just incase. the system has also been working for months untill a few days ago....i am using the Netborder Gateway with a Sangoma card.

I dont think it is a wireshark issue as the logfile does not show that it is playin Ringback.wav...i will show below the logfile. 5] 2009/02/05 09:18:47: Identify trunk (IP address and DID match) 2 [7] 2009/02/05 09:18:47: Set packet length to 20 [6] 2009/02/05 09:18:47: Sending RTP for e74464f9-f38f-11dd-bf57-d03f7f763324@pbx#ae480c5bb3 to 10.0.10.16:14010 [5] 2009/02/05 09:18:47: Trunk T1 sends call to 8401 in domain localhost [7] 2009/02/05 09:18:47: Attendant: Calling extension 8401 [7] 2009/02/05 09:18:47: Set packet length to 20 [7] 2009/02/05 09:18:47: Call e05e8ec6@pbx#3380: Clear last request I HAVE LOOKED AT OTHER PBX'S AND THEY SHOW THE FOLLOWING LINE WHICH THIS PBX IS NOT SHOWING [8] 2009/02/05 09:20:32: Play audio_en/ringback.wav why would my phone system stop playing this back? system files corrupted? pls advise asap thank you

Hi; it was using Message 180 at first but even after i changed to media it still didnt work!!

HI; i am using a PRI card with pbxnsip latest version but i have an issue where a caller calls in and does not hear any audio untill the caller picks up or it goes to voicemail...im assuming it is not playing the ringback tone but i checked the audio folder and the ringback.wav file is there. pls advise asap as clients are hanging up the phone because they think it is not going through

1) i have way more than 10 extensions...are you saying there is no way to do this? is there a plan to implement this? this is going to be a big problem for the client. 2) a/a??????

Hi; i have two PBXnSIP server trunked together for a company that has one office in NJ and one in NY. although internal callers in each location are able to dial the extension of the other office without a problem i do have a number of issues. 1) when a caller calls in from the outside and reaches the auto attendant in NY and they try to enter a NJ extension it tells them "this extension does not exist" is there any way to setup the possibility of callers being able to do that without a problem? 2) when i tried to use the hot desking feature from one pbx to the other it does not work but when i forward all calls it works fine...is there any reason for this and how (if possible) can i resolve this? please advise. Thank you

HI; i tried that but still get the same error message.

After reading into this further i found that there is supposed to be a global seeting "offer pickup" that can be turned on or off but i cannot find this setting...i have seen it before but do not know where i is! any help would be greatly appreciated!

Hi; it seems that a lot of these issues have been fixed in the latest versions of pbxnsip however, i have two problems; 1) It seems that i am not only monitoring the calls but any call that comes in to one of the extensions i am monitoring are ringing on my phone too. 2) I cannot park a call by just hitting the call park button.\ Number One issue is way more severe as the partners dont want the receptionist picking up calls that have been routed directly to their phones. is there any way to toggle that setting? pls advise

HI Can you advise what issues these files fix? thank you

The given path's format is not supported If it makes it easier, I’m including the code that I’m using to connect. I’ve connected with SOAP using similar coding, but ever server is different: -- Dim strXML As String = "" strXML &= "POST /soap.xml HTTP/1.0" strXML &= "Content-Type:application/xml" strXML &= "Content-Length: 400" strXML &= "<?xml version='1.0' standalone='yes'?>" strXML &= "<env:Envelope xmlns:env='http://schemas.xmlsoap.org/soap/envelope/'>" strXML &= "<env:Body>" strXML &= "<sns:CreateTrunk>" strXML &= "<Domain>sip.jivetel.com</Domain>" strXML &= "<Name>WinnerTest1</Name>" strXML &= "</sns:CreateTrunk>" strXML &= "</env:Body>" strXML &= "</env:Envelope>" 'Send Dim Xml_Returned As String Try Dim myWebClient As System.Net.WebClient = New System.Net.WebClient Dim bytes() As Byte = myWebClient.UploadData("XX.XX.XXX.XXX:8080", "POST", System.Text.Encoding.UTF8.GetBytes(strXML)) myWebClient = Nothing Xml_Returned = System.Text.Encoding.ASCII.GetString(bytes) bytes = Nothing Catch ex As Exception Xml_Returned = ("<strong>Error returning xml:</strong> " + ex.Message) End Try please advise.

so what is the resolution?

I’m writing in ASP.NET and sending a simple XML file to set up a basic trunk: strXML &= "POST /soap.xml HTTP/1.0" strXML &= "Content-Type:application/xml" strXML &= "Content-Length: 400" strXML &= "<?xml version='1.0' standalone='yes'?>" strXML &= "<env:Envelope xmlns:env='http://schemas.xmlsoap.org/soap/envelope/' xmlns:sns='http://www.pbxnsip.com/soap/pbx'>" strXML &= "<env:Body>" strXML &= "<sns:CreateTrunk>" strXML &= "<Domain>sip.jivetel.com</Domain>" strXML &= "<Name>WinnerTest1</Name>" strXML &= "<Type>register</Type>" strXML &= "</sns:CreateTrunk>" strXML &= "</env:Body>" strXML &= "</env:Envelope>" but i am getting a response "The given path's format is not supported" can anyone help me and tell me what i am doing wrong? thank you;

Hi; i am opening this string on the forum to document every step of my attempts to use SOAP to update the system in every way. i will post comments suggestions and questions here so please stay tuned so that we can get this working together. thank you;

HI; ever since we upgraded to versino.3120 any grandstreams phone in our enviornment are unable to send dtmf in voicemail menu...DTMF works when calling other phone numbers. please advise asap. thank you

Another thing you guys might want to think of is i know that you are planning on adding functionality for multiple cell phones to be entered in that section but are you also going to add the option to have to "press one for confirmation" for each and every one?

so the ivr node would be able to redirect a * prompt to any destination?

HI; i have a Polycom 550 with PBXnSIp which is working great configured through PNP. however i consult for a second company and would like to be able to register my extension from second server on the same phone. now i am aware that only one phone can be configured per extension and one system per phone using PNP but even when i try to manually add settings through the GIU or through the phone itself it never seems to be able to register. does anyone have any experience with this to save me the trouble of having to spend hours on it? please advise. Thank you;

i could tell them to get over it but before i do that all i want to try is to take out 8 from the extensions in the dial plan...meaning that right now when the phone dials 2-9 it is only looking for 3 digits...i want that when it dials 8 it will require the # and wont auto dial the number (like it does for extensions beginining with 1) please advise. thank you;

i have another idea...please advise if this would work. basically currently the issue is that when the user enters 8121 the phone does not recognise 1 as a prefix to an extension and therefore cuts off the fourth number because the dialplan states that 8 is a prefix to an internal extension and therefore the phone is saying send this call to 812. however if we took 8 out of the dial plan (there are currently no extensions beginning with 8) then maybe the phone would wait for the fourth digit (in this case 8121) and when the receptionist hits # it would work? for testing purposes can you please tell me how i can go about removing the 8 i(ie the dialplan should state [2-7]??

can anyone help me out a little? i am logging in with psftp but cannot find the firectory where all the wav files are saved. please advise. thank you;

Hi I have a client who already has his AA's recorded and has a prompt in any sub AA's that give the caller the option to press * to be redirected to the Main Menu. however when the caller presses * nothing happens...is there anyway to allow this to work? pls advise asap.