Sipdroid with Trixbox/Asterisk – How????

I have spend many hours trying to get my ADP1 phone running Sipdroid to register with my trixbox server. Its driving me nuts, really, it is.

Yes, I have changed the sip_nat.conf file and ensured it has…

nat=yes
externip=203.blah.blah.blah
localnet=blah.blah.blah.blah/255.255.255.0

I have tried it over WI-Fi on my LAN and over 3G. I was able to reguister OK – once then if fails there after. Strangely it will register only after you creat a new extension for sipdroid in trixbox.

Is it my NAT router doing something odd?

So, do you think I can get it to work. No, I can’t and its driving me NUTS! AHhhhhhhhhhhhh.

I think I’ll just buckle to it all and register with PBXes and then register a VSP as a trunk there and use it that way. Surely it couldn’t be this difficuly but apparently it is. Interestingly on the Sipdroid site it says something to the effect, we are smarter than you and have changed PBXes a little from a standard trixbox installation (obviously) so don’t bother wasting to much time trying to get Sipdroid working with your own asterisk box – it won’t work. This really shits me to be honest. Shouldn’t these kind of things be created to give the maximum amount of interopability.

So who has managed to get Sipdroid register and work with your own Trixbox/Asterisk server? Let me know how you did it – its killing me.

If someone at PBXes reads this – what changes need to be made to my Trixbox to get Sipdroid to work with it, come on guys…..

8 Replies to “Sipdroid with Trixbox/Asterisk – How????”

  1. I managed to get it working without too much difficulty. The SIP account setup is probably the most important. I’m running Asterisk 1.6 (no Trixbox), and here are snippets of the config:

    sip.conf
    [android]
    type=friend
    regcontext=internal
    regexten=100
    callerid=”android”
    host=dynamic
    allow=all
    context=internal
    qualify=yes
    secret=1234
    nat=yes
    canreinvite=no

    Then setup sipdroid to point to the server, with the proper username/password and you should be good to go.

  2. Thanks, for this, I’ll give it a whirl and see how I go.

    I presume you also had entries in your sip_nat.conf file (assuming you have such a think in vanilla Asterisk 1.6)?

    Are you using TCP for sip signalling or UDP still? I recall somewhere that Asterisk supported it now.

  3. That doesn’t work for me. I’ve tried all the possible combinations:
    – asterisk 1.4
    – asterisk 1.6
    – udp transport
    – tcp transport
    – nat
    – trixbox
    – PBX in a flash
    – 32 bits
    – 64 bits
    – different kernels
    – different sipdroid versions (the latest I’ve tried is 1.1.0 beta posted 4 days ago)
    – using htc hero with latest version

    There’s no help anywhere out there… 🙁 quite frustrating! This guys really want you to use their undocumented/unsopported pbxes.org service.

    1. Yeah. I like it because it works using pbxes but it annoys me that the thing doesn’t work the way it should – ie with any asterisk installation.

      It also may not be working because of network or asterisk config. I wonder if the success stories are from those who don’t have it behind a NAT router?

      I wish the development would concentrate on asterisk support and a lower bandwidth codec but given I’m not a coder there is nothing I can do. Its annoying as this appears to be the only sip client out there for android

      Competition is needed I think.

  4. I feel your pain. I am in EXACTLY the same situation. I tried Elastix, Trixbox, PBXinaFlash, AsteriskNow and my own install compiled from scratch. Asterisk v1.4.x and v1.6.x. Always the same thing so it’s a fundamental problem.

    My server is in a datacenter with a public IP and no NAT so although NAT on the sipdroid end is likely the cause it is not the problem. I have SIP phones and sip client software like Xlite behind routers and they don’t have any problems. Just sipdroid. You can open up the relevant ports on the router on the sipdroid end and it will work but that is not a solution.

    So working backwards, it’s not a sip_nat.conf issue and not an Asterisk issue in general. Qualify=no does not solve the problem. All that does prevent Asterisk from checking if the phone is there. So Asterisk will contact the phone when you dial that extension but it will still ultimately fail.

    About the best explanation I came up with is that Sipdroid does not continually re-register and ping (not sure the proper SIP term) the server which keeps the incoming ports open on the router. The explanation from the developers (I believe) was that this was done to extend battery life. This should be optional and not hard coded.

  5. What did it for me was changing the protocol in SIPDroid to UDP from TCP. Counter Intuitive but now it works with standard extension setup from TrixBox 2.8 (asterisk 1.6).

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