Friedom-Tech
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Posts posted by Friedom-Tech
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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.
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?
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Well, the reason could be that the system is not supposed to signal that the other side is actually ringing. Or something simple like the log level is set to 7 and the PBX would not show a message on log level 8. I assume the file is in the file system. Corrupy file systems are pretty low probability these days.
It could be really something that the phone is really not ringing or that the PBX does not know about it. A Wireshark trace from the PBX server will show this.
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.
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make sure you downloaded:
http://pbxnsip.com/download/audio_moh_30.zip
and
http://pbxnsip.com/download/audio_en_30.zip
are you using the NETBORDER product? or a gateway?
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.
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Looks like you need Wireshark to see what is going on.
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
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Some trunks cannot deal with early media. Check if you have the flag "Ringback" set to "Message 180".
You can easily see if the PBX sends media by using Wireshark. That should further help to nail the problem.
Hi; it was using Message 180 at first but even after i changed to media it still didnt work!!
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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
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1) Yea, that is a common problem in this kind of setup. As long as you have less than 10 destinations you can use the direct destinations feature to work around this.
2) Hot desking is like changing the registration. The call redirection is sending out new calls. That is why hot desking cannot work across different PBX.
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??????
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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
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Looks like the SOAP message is not even reaching the PBX.
Just to verify, try IP address instead of sip.jivetel.com. Also, I am assuming PBX is running at XX.XX.XXX.XXX:8080 (fyi... port 80 is the default port)
Is there any detailed error on the
HI;
i tried that but still get the same error message.
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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
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!
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I recently installed a system with Polycom 550's and want to take advantage of the 3 softkeys I have on those phones. I tried to think of a good way to emulate a key system since I was taking the customer away from an old Avaya system , what I cam up with was this ... I created 3 park orbits 801, 802 and 803 and add them to be monitored extension on all the extensions. This made them populate to the lines 2,3 and 4 respectively on the phones ,
items I was happy to discover ,
1- I could make a quasi line 1, line2, line 3 etc for them in sort of a key system style , using co1 co2 etc wouldn't work since I could only monitor 3 lines and they had around 30 phones .
2- I could use the *85 and it would automatically read me the extension the call got parked to
3- the BLF lights indicated on all phones a call was parked on an orbit
items I think we could improve to increase sales ..
1- I would like to be able to push the corresponding lit button to pickup the call rather than dial *86801 ... this would be a cleaner approach and would be nice since the customer's first instinct was to push the button to retrieve the call ..
which leads me to problem 2
2- if you push the button it calls the extension you used for the park orbit , if you forget to turn off voicemail it asks you to leave a VM for extension 801 ... this seems an unprofessional and makes the feature seem as a work around ...
so in a nutshell , can we create a raw extension type that for Park Orbits and can we make a way to allow those parked calls to be picked up by dialing them directly or using the buttons on the phone to retrieve them , it seems dialing extra digits is simple but customers coming away from key systems would be very happy to use this , especially considering the functionality is almost 80% there ..
y
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
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For anyone interested, I've attached an updated polycom_sip.xml. This is for the Polycom SIP 3.1.1 release (use in conjunction with BootROM 4.1.2). I took the parameters from pbxnsip 3.0.1.3023's built-in polycom_sip.xml template and merged them with the new sip.cfg that comes with Polycom SIP 3.1.1. It appears to have corrected a few issues I was having but the settings should be tested before putting into production.
The file needs to be placed in the /html directory (on Windows this would be C:\Program Files\pbxnsip\PBX\html) and then reboot your Polycom phones. To revert to the original settings, just remove this file from the html directory and reboot the Polycoms.
We also found an error in the existing polycom_phone.xml (missing space). It causes problems with outbound dialing. The corrected version is also attached. It also goes in the /html folder.
HI Can you advise what issues these files fix?
thank you
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Take 'xmlns:sns='http://www.pbxnsip.com/soap/pbx' out and try.
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.
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Maybe it really checks the www.pbxnsip.com URL. There is no document
and maybe that is a problem for ASP.NET.so what is the resolution?
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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;
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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;
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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
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i think it is a nice feature because not always do you want to pick up calls from the office but if you dont know who it is you feel obligated to pick it up.
i would say try doing two things:
1) give the option to turn that on or off ( the users who are pushing ANI it is a waste)
2) speed up the playback a little its currently at snails pace!
let us know!
Thank you for all the great additions!
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?
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The star key usually has the function to clear the input; that is probably the problem here. I believe at the moment there is no way to get the AA to redirect a call when entering star. Workaround would be to use a IVR node.
so the ivr node would be able to redirect a * prompt to any destination?
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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;
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The [2-9]11 is essentially for 911. Maybe you just write 911; maybe also include 411 if someone wants to dial it. This way you can take the 811 out.
In the end, you need to test test test. IMHO life would be easier if they can move the 1xx extensions out of the way or just accept that they have to dial 9 for an outbound call (to "seize" a line, haha). If you have to tailor them their own dialplan that will mean some work.
Or they get over the whole thing and press Dial after entering a number. Thanks to the several billion cell phones out there that is not such an crazy idea!
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;
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Well, if you want that people can dial extensions starting with "1", then you can not use the "1" as the prefix for national numbers (11-digit dialling). You can tell customers to use something like "91xxxxxxxxxx" or "9xxxxxxxxxx" (9 would be for "outbound line"). However this will bring trouble with the address book, as address book entries on the phone will not start with "9" in the beginning. And of course you need to change the dial plans The pattern for this would be below.
[1-7]xx|8[1-7]xx|[2-9]11|9,1xxxxxxxxxx|9,011x.|*x.
You can change that in the dialplan.xml file attached above. The comma AFAIK will give you secondary dial tone.
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]??
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You can also use psftp on Windows (which it putty), no need to learn shell scripts in Linux!
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;
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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.
and maybe that is a problem for ASP.NET.
NO RINGBACK TONE
in General Setup
Posted
please see attached wireshark trace....im trying to attach it!!!!doesnt allow me to!!!