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Vodia PBX

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Posts posted by Vodia PBX

  1. We have a patton 4114 gateway and 4 pstn lines.

    1 of those is ocassionally used by our fax machine. (a true fax machine someone can walk up to)

    We need to keep the fax machine.

     

    Will I fiddle a lot or can i get a t.38 fxs/ata and have the AA detect and forward to fax machine?

    Also outgoing faxes?

     

    Would be nice but I don't want to fiddle and fiddle. Also, does someone have ATA/fXS they recommend?

    (I would use a patton m-ata, if no suggestions)

     

    We all know that T.38 is not a masterpiece with regards to simplicity and interoperability and features like encryption are still missing. If you are a pragmatic guy and your business cards printed a DID number that correspond to a specific cable coming out of the wall - yes I would plug that straight into the FAX. Or if you are a real pragmatic guy and you are running the PBX in the LAN you say so what and skip the whole T.38 part and just use G.711 to the FAX. The LAN does not loose packets.

     

    The only case where you really need T.38 is when you want to run the FAX over a line that does drop packets, e.g. when using a ITSP service or having the FAX in a remote office. Then you have no big choice - T.38 is the way to go. Hopefully your ITSP supports T.38 already.

  2. exactly, id like to see a "i am stupid and forgot my password so please email it to me"

    but like i said, it should be based on per user, some users can do it, and some should be blocked

     

    Yes, plus we just added at least the initial email with the web password. If the user is smart enough to go through old emails he might find it. Probably we should add a button that triggers the sending of this again. But still be must mandate that the user enters a valid email address - which is a challenge for some...

     

    Stupid end users! :rolleyes:

  3. Is it possible to create a url that will include the users account and passwords to ease their login?

     

    I.E. a Desktop ShortCut Named PBX with the address of http://192.168.1.5 /user=201,pw=1234

     

    maybe the password is the encrypted version, but would need to be accessed on the users web portal to create the link...

     

    Maybe a button in the user interface to create the entire link to be copied to the desktop would be better.

     

    Yea, I know the authentication is a pain. The first thing users can do is to set a permanent cookie (during login). That should help easy the pain. For some things, the above mechanism is already available (e.g. see http://wiki.pbxnsip.com/index.php/Netsuite).

     

    We are looking into other mechanisms like client certificates. This would be the right way of establishing trust relationships between the end user and the PBX and this would also make sure that the traffic between PBX and user is safe.

  4. How do I "unlock" that file?

    Seems like that limits the ability to give real/or semi-real time stats on the phone system?

     

    The next day it will be free... But the next version will not lock it any more. Then you can read it anytime - at the risk of reading an unfinished write at the end of the file.

  5. I have a question regarding SRTP/SDES. Is it possible to configure pbxnsip to use SDES with a SRTP authentication tag length of 80 bits (AES_CM_128_HMAC_SHA1_80) by default? Because of what I see, it uses a tag length of 32 bits (AES_CM_128_HMAC_SHA1_32), even if 80 bits is supported for incoming calls.

     

    Hmm. No that is hard coded. IMHO the 80 bits are just overkill, why would you need it?

  6. That response code usually reports about trouble with the codecs. Typically it means that there is no common codec that both sides can agree upon. However, in the case above, it does include G.711 and I can hardly believe that they don't support that codec. My second guess would be that they expect a certian packet length; however we don't insist on anything and it should be okay. Hard to say what is "not acceptable here".

  7. NO, it does not get idle. It just stops indicating correctly.

     

    Testing this requires only one xlite. Setup the xlite correctly, add YOURSELF as a contact. (change the availabliity and watch it change down in the contacts to make sure it works) Change it a couple times to verify to your self it works...now wait several minutes and change it....it will not change down in the contact list.

    (watch 3-5 minutes)

     

    I'm not sure is dropping the ball: us or xlite?

     

    On presence:

    For remote users this is not micky mouse...it can save time...why bother calling, waiting till vm answers if the person isn't even there or on the phone? Same with IM, why IM if he's out shoveling snow? ;-)

     

    Since companies are installing servers just for internal IM presence is something they want. And I think we are pretty close to having it.

     

    Just updated http://wiki.pbxnsip.com/index.php/Counterpath.

  8. will it have a forgot my password tool? and also be able to set per extension if it can have access at all? if not the forgot my password wont work

    so lets say company 1 doesnt want extension 111 to have access, but if it does a forgot password, thenit will get the password, which isnt good

    so for extension 111 have the option of not checking on access to web portal

     

    No, this is just about the HTTP/HTTPS socket. You can limit the access by choosing password that the end user does not know. Ouch, don't do that on 3.1.2! But it will work on 3.2.

  9. for now we have installed IPTABLES on the server and we block out all IP except our office, so we wont have a problem with customers tryign to log in with a wrong user name or password match

     

    You can also bind the http server only to your private IP address. In the port settings, put something like "192.168.1.2:80" and then it will bind only to that port...

     

    A new version will be available today.

  10. When pressing * or #, the AA doesn't accept these user inputs for redirection to another AA (for example).

    But if I replace with a number instead of * or # it works fine, and the call is re-routed to a destination.

     

    Yea, that was a design decision ("feature"). Star usually means "clear", this is the way people can correct their input. And pound usually means "enter", which e.g. is useful when customers are using extensions with variable length.

  11. i thot it was after xlite presence idle timeout, but i set it to 1440 minutes and still stopped showing presence after several minutes...

     

    You mean because the phone gets idle? That sounds like a feature to me. IMHO the whole hype around presence is completely over. I understand IM is useful, but I never understood what presence should be good for. My productivy dropped like a stone when people saw that I became online. Hiding my presence is a feature! :)

  12. I've done this but I can't access the resulting file because it is always in use.

     

    Maybe I'm doing something wrong?

     

    Ah, you are probably a victim of the Windows file locking mechanism. Seems like we have to Open and Close the file all the time...

  13. 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?

     

    Yea, that is the intention. When you perform a attended transfer to the cell phone, you will be able to hear if the other person picks up or the mailbox. There are other situations (e.g. when the PBX call the cell phone because there is a new voicemail message), but in all these cases the user has to interact with the PBX before the interaction happens.

  14. 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?

     

    Yes you can. All you need to do is use a symbolic link to another destination.

     

    We start to support SQL also from the PBX. Send an email to support@pbxnsip.com to get more infos on that.

  15. About the more "theorical" questions, I'm curious about the reason why I see UDP packets during conversations when pbxnsip is configured for TCP, I thought it used RTP over TCP...

     

    Nonono. Audio is always sent over UDP. TCP has huge delay if a packet gets lost. Instead of repeating a lost packet (and letting all other packets wait) you better just play back a little click and then the audio will go on.

  16. 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?

     

    Are you using failover in the trunk settings? There was another post today about that topic, see http://forum.pbxnsip.com/index.php?showtopic=2039. Maybe that solves the problem.

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