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Posts posted by markosjal

  1. I have spent weeks configuring and testing many possible streaming solutions and I find it is easier to post here my needs rtather than configure and test another"solution" that will not work for me.



    Will this product allow me to connect from a remote location ,(camera) to the server? In other words will it sit and listen for an incoming connection, and when it hears it use it as the source for the stream? I am referring much to the way that a shoutcast server sits and waits for an incoming connection. What would I need at the camera end to do such a remote signal to the server?


    Will this product stand by and play a file or image while there is no live remote camera?


    If we buy gthe full product, Can we use our existing JW player for the files to ensure consistency on the web site?





  2. I have determined absolutely that no matter what I do, uplinj seems to send the Hamachi Network Address Adapter address to my asterisk whenever Hamachi is Active. I imagine in many cases this nmay also occur with multiple NICs installed. IN my case there is one physical NIC, one Hamachi Virtual NIC , and one Virtual LAN v(ia Virtualbox)


    I have tried various options and a "sip show peer XXX" in Asterisk shows that Hamachi Network adapter address. This means my asterisk is trying to send to a hamachi IP and not a real IP. I suppose I could install hamachi on my asterisk, but that would be a work around., and may be unrealistic for many users.


    In fact I am trying to use a private LAN IP from the machine running Uplink to the asterisk box. I have tries also with the routeable public IP and the result is the same.


    I tried this about 1 year ahgo on my desktop computer and identified the Hamachi incompatibility so I think it is decisively the problem



    I really would like to know how many of these one way or no audiodio posts are by users that are using hamachi.

    , and to know if anyone has found a solution to this. .


    I think this is a major error for NCH to make especially given there is a field to fill in for a fixed IP that seems to not do the job.any users may find this bug only after paying to register and it is not right.

  3. I can see that some say this is possible , but when I try to run another instance of uplink as another user on the same desktop, the existing uplink window simply reconnects to skype.


    So I logged into the machine as another to start the second instance, user over Remote desktop and here is what I foud. Both Skype and uplink run happily as a second instance, however uplink insists on

    1) a new license

    2) registering to the same account, even though it automatically chooses the next SIP port.


    I was hoping that since they are two different accounts i could truly direct the SIP calls differently.


    If NCH can separate the Registration info (serial number) why can the 2 instances not register to different SIP accounts.? If I change the SIP account info in either instance it changes on both instances, again with different SIP ports and this makes no sense.


    I believe this is a programming oversight.

  4. I'm using asterix Trixbox. I finally got uplink to register with my Trixbox server. When I try to dial out I get the ringing tone, and the uplink console says incoming SIP call, but it never makes it to the destination. Somewhere between Uplink and Skype is the problem.


    I'm using uplink ver1.30 and Skype ver3.8.0.188




    Some of you guys need to post more information, specifically how uplink is configured, in the section about what to dial on Skype , and a SIP trace from Asterisk is always helpful as well. If peopl guess at your problem they can waste a lot of everyone's time.

  5. I am looking for a software package that will allow me to put a delay on outgoing voice.


    We do live radio interviews on the streets of Key West Florida and must be prepared to cut off profanity with the party goers.


    Do you have anything like this?




    Do you know where I can get such?


    Thank you in advance for your help.


    This is where a good old fashioned tape loop comes in handy!

  6. Still have this problem with calls coming from zaptel. Anyone has a fix ??? Sip is perfect


    More than likely this is because asterisk 1.2 has no real jitter buffers. This may affect your connections are far from the internet backbone.


    You may be able to minimize this effect by setting up reinvites. canreinvite = yes, among other prerequisites. This depends much on your configuration however which I know little about.

  7. FYI, I do this in FreePBX with a custom trunk (that points to an extension) like this

    Custom trunk = SIP/$OUTNUM$@4999

    where 4999 is the extension that SIP2skypre registers to.


    I then establish extensions for those users that I must reach by skype. In the dial command on the extension I then use SIP/SkypeUserName@4999, and uplink merrily sends the call to skype.


    to clarify each extension serves as a skype destination. I could just as well do it this way in the extensions SIP/SKypeUserName@uplinkIP:5070 but I think that what I defined previously is better because it better accommodates an IP address change.



    It is cumbersome, but the best solution I have found for the people I must call on Hype.

  8. I am hoping that someone may be able to properly decode this file and tell me how I may make an equivalent file in the G711u and G711a formats,


    I have tried several utilities to plan and identify these files, however I can safely say that none do. Most utilities do not properly identify it, ands most players do not play the voice just a noise. This file is a voice, not a noise.


    I do know this is some g729 format






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