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  1. #1
    Master Untangler
    Join Date
    May 2008
    Location
    Orlando, FL
    Posts
    105

    Default VOIP Phone Server behind Televantage. Untangle = fail

    I can't seem to get my Televantage VOIP phone system to work behind Untangle...

    Well, it can make SIP outbound calls just fine to the ITSP. That's all we personally use it for.

    BUT.. I have cases where I need SIP clients off our network to register to the server as external extensions.

    For the life of me, I can't get them to register. I even manually port forwarded 5060 and RTP ports. That has worked with even $50 linksys firewalls in the past. I turned off ALL racks in Untangle, still no go.

    Yes, untangle is VOIP friendly if you're connecting to an off-site server using a Softphone or Hardphone.... but I'm really pissed off at this crap that it doesn't work if the SIP server is behind Untangle and people are trying to connect to it offsite.

    They need to specify that in the manuals and advertisements. VOIP Friendly, as long as you're not the one with the server.

    I've been using untangle now for over a year. If I had to start all over, I would have just bought an SSG 20. Untangle is a great idea, but it's still years behind the competition.

    /rant

  2. #2
    Master Untangler
    Join Date
    Oct 2008
    Posts
    141

    Default

    First of all i would connect via VPN. That would solve it.

    I guess you are using NAT? SIP and NAT sucks. I have gotten it work with asterisk pbx though. I had to manually specify my public IP in the asterisk SIP configuration. Then it did work. I don't know your PBX solution, but it probably should have a setting like that.

    In untangle you could try enable or disable "SIP NAT Helper". It's under configuration > Networking> Bypass rules (advanced)

  3. #3
    Untangle Ninja sky-knight's Avatar
    Join Date
    Apr 2008
    Location
    Phoenix, AZ
    Posts
    26,241

    Default

    config -> networking -> advanced -> bypass

    Untick the box that says "enable SIP NAT Helper"

    Try again... your PBX more than likely is configuring itself to adjust the SIP Packets for NAT... if you do that twice it's a no go.
    Rob Sandling, BS:SWE, MCP
    NexgenAppliances.com
    Phone: 866-794-8879 x201
    Email: support@nexgenappliances.com

  4. #4
    Master Untangler
    Join Date
    May 2008
    Location
    Orlando, FL
    Posts
    105

    Default

    Nope, still no go. As I said. It works fine behind sonicwall, linksys, etc.

    It's similar to an asterisk, but more gui and more expensive.

  5. #5
    Master Untangler
    Join Date
    Aug 2008
    Posts
    970

    Default

    Whats the NAT setting on the phone in your televantage?

  6. #6
    Untanglit
    Join Date
    Mar 2009
    Posts
    23

    Default

    My IPPBX works fine behind Untangle, and with remote users as well.

    Here are two scenarios:

    phone-public internet-untangle-ippbx - Should work pretty well, long as you specify the public IP to the ippbx

    phone-nat-internet-untangle-ippbx - Phone needs to support STUN, otherwise it will report incorrect (private) ip to IPPBX, and communication will fail. I ran into this with my otherwise fantastic Polycom phones. They don't support STUN.

    Before giving up on Untangle, try working with a softphone that supports STUN, like xLite, and narrow the problem down further. Things SHOULD work.

  7. #7
    Newbie
    Join Date
    Jun 2009
    Posts
    1

    Default VoipSipSdk

    VoipSipSdk

    I am now looking for voip solutions. And found information about Voip sdk.
    According to their website
    www
    voipsipsdk
    com
    Voip sdk is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, OpenSER other.

    They have all features I need:
    # Dynamically loadable codecs
    # Registrar support
    # Play wav files into conversation
    # Record conversation into file
    # Hold/Retrieve call
    # Forward Call (Blind Call Transfer)
    # Transfer Call (Attended Transfer)
    # Mute Sound
    # VPN support
    # Noise reduction
    # Auto gain
    # Jitter buffer parameters
    # Samples on Delphi, C#, VB, VB.NET, C++ 2005, C++ 6.0, HTML (SIP ActiveX)
    # Windowless samples on C++ and .NET
    # DTMF
    # Adaptive silence detection
    # Adaptive jitter buffer
    # STUN support
    # Comes as ActiveX control

    But before I will download the evaluation version I would like to hear other people experience.

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