Integration with Fring

  • Dear Starface Team
    I'm trying to make fring on iPhone ( work with starface.
    Fring has a SPI module, for some SIP providers it is preconfigured, but there is an "other" category where one can simply enter the SIP server name/address username and pass and try to connect.

    This fails with an "invalid username or pass" message, although I do not believe this is the problem.
    I contacted Fring support but so far they did not come up with a solution.

    You probably do not have an answer to me right away, but could you please tell me how can I see whether the iPhone arrived to the SF server at all or not.

    The asterisk -vvvv does not show unsucefull logins. I tried it with my regular SIP client and if I enter bad username or pass the sterisk -vvvv shows nothing on register attempt (it does show if the client registers sucesfully, though)

    So where could I look for some info on unsuccessful login or SIP client connection attempt in general

    Thanks in advance

  • Hi zmagyar,
    what Login-Data are you setting within the fring client? Be sure to set the data you set in the telephone-proilfe of the starface. Not the login data that you use to log yourself into the Web-Interface.

  • Sure, I created a separate phone for the iPhone. It is also a question which SIP phone type should I set, i found some info on their support pages about which codecs and such should be used, but it still does not work.
    As said, to start to troubleshoot, I would like to see first if the login request reaches the SF at all or not. Once I know this maybe we learn something more from the logs.
    BTW it is a cool setup for SF users because it makes you use your mobile phone as wireless SIP phone inside the office, so I guess it would be a nice thing for you too to make it work. iPhone is a hot topic nowadays...
    Thanks for your help (BTW you can talk to Phillip, he was working to me during the initial SF setup, he should remember us ;-))

  • Hi zsolt,
    in general you can use the "standard sip" phone profile in the starface. If you know any specifications Fring needs, you can create your own telephone profile.

    In order to debug the connection to your starface you can connect yourself via SSH to the STARFACE and go to the asterisk console.

    asterisk -rvvv

    Here you can focus the debugging on the IP adress of your iPhone.

    sip debug ip x.x.x.x

    Now you will see every SIP package going to or coming from your iPhone.

    Edited once, last by Torsten ().

  • This is what I did, I was just missing the sip debug ip command, will try tomorrow, today I'm in home office

  • Hi Torsten,
    I tested it, if I enable debug then indeed I get some input if the username/pass is incorrect. I will paste it FYI below, but is not relevant for this post.
    I tried it then with the Fring client and it returned no output, so I would say that it does not even arrive to the SF server.

    What they said before is this:
    "Since we do not rely on the handset's SIP capabilities the SIP traffic is routed through our servers.
    Therefore, your server has to be accessible from the net and not only internally."

    So I guess, once I click on register on the fring client, it goes out to the internet, to their servers, does something and then (again I guess) tries to reach the SF server thru public IP...
    We did forward the port 5060, but still no luck. I guess it is them who need to answer this. But I'm reporting to you, just in case you have a wild guess.
    Thanks for help so far

    I tested it with my regular SIP client and this is what I get (just FYI):
    <-- SIP read from
    REGISTER sip:;transport=UDP SIP/2.0
    Via: SIP/2.0/UDP;branch=z9hG4bK-d8754z-6255647451b1d32b-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:zsoltXllito@;rinstance=465512e7f1d255cc>;transport=UDP
    To: "zsoltXllite"<sip:zsoltXllito@>;transport=UDP
    From: "zsoltXllite"<sip:zsoltXllito@>;transport=UDP;tag=7d7f4e14
    Call-ID: M2Y4MDM3NzllMWZjMTMwMzY1MWViNTNiMjk1MDk5ZjI.
    CSeq: 1 REGISTER
    Expires: 3600
    User-Agent: Zoiper rev.1809
    Allow-Events: presence
    Content-Length: 0

    --- (13 headers 0 lines) ---
    Using latest REGISTER request as basis request
    Sending to : 5060 (NAT)
    Transmitting (NAT) to
    SIP/2.0 404 Not found
    Via: SIP/2.0/UDP;branch=z9hG4bK-d8754z-6255647451b1d32b-1---d8754z-;received=;rport=5060
    From: "zsoltXllite"<sip:zsoltXllito@>;transport=UDP;tag=7d7f4e14
    To: "zsoltXllite"<sip:zsoltXllito@>;transport=UDP;tag=as579b2f2f
    Call-ID: M2Y4MDM3NzllMWZjMTMwMzY1MWViNTNiMjk1MDk5ZjI.
    CSeq: 1 REGISTER
    User-Agent: STARFACE PBX
    Content-Length: 0

    Scheduling destruction of call 'M2Y4MDM3NzllMWZjMTMwMzY1MWViNTNiMjk1MDk5ZjI.' in 15000 ms

  • Good news, it works (half), I had to enter the public address of our SF server not the internal one, and then it works.
    One thing to go, though, the Fring client (the caller) can hear the other party, but the other party (the called one) does not hear nothing. I guess this must be some port not forwarded thing.
    Any idea?

  • Hi zsolt,
    interesting solution from fring, routing all the traffic through their own servers. I wonder what they are logging on these servers. :eek:

    Anyway, ....
    the problem you described:


    the Fring client (the caller) can hear the other party, but the other party (the called one) does not hear nothing.

    This sounds like missing forwards for the ports 10000 to 20000 for RTP. Once these ports are also forwarded to the STARFACE both caller and called person should hear each other.

    Best regards,

    Edited 3 times, last by Torsten ().

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