symbian fails over an E65

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Hi Rodrigo,

I don't think the config has something suspicious,
as I said previously, perhaps you can see the stack trace (SIP, media,
etc) in asterisk,
or try to make direct call with pjsua on desktop while log level is set to 5/6.
If on both cases the assertion still occures, you can report more
details on the reproducing steps.
If only one of them is rising assertion, you can compare the traces produced.

nanang


On 13/02/2008, Rodrigo Vega <vegaroy13 at gmail.com> wrote:
> Alright...
>
> I don't know Nanang if you can check out if my configs are alright, let me
> know if you notice something strange:
>
>
> > //
> > // Basic config.
> > //
> > #define SIP_PORT    5060
>  >
> >
> > //
> > // Destination URI (to make call, or to subscribe presence)
> > //
> > #define SIP_DST_URI    "sip:echo at 192.168.1.73"
> >
>  > //
> > // Account
> > //
> > #define HAS_SIP_ACCOUNT    1    // 0 to disable registration
> > #define SIP_DOMAIN    "192.168.1.73"
> > #define SIP_USER    "symbian"
>  >  #define SIP_PASSWD    "symb"
> >
> > //
> > // Outbound proxy for all accounts
> > //
> > #define SIP_PROXY    NULL
> > #define SIP_PROXY    "sip:192.168.1.73:lr"
> >
>  >
> > //
> > // Configure nameserver if DNS SRV is to be used with both SIP
> >  // or STUN (for STUN see other settings below)
> > //
> > #define NAMESERVER    NULL
> > //#define NAMESERVER    "192.168.0.1"
>  >
> > //
> > // STUN server
> > #if 0
> >     // Use this to have the STUN server resolved normally
> >  #   define STUN_DOMAIN    NULL
> > #   define STUN_SERVER    "stun.xten.com"
>  > #elif 0
> >     // Use this to have the STUN server resolved with DNS SRV
> > #   define STUN_DOMAIN    "iptel.org"
> >  #   define STUN_SERVER    NULL
>  > #else
> >     // Use this to disable STUN
> > #   define STUN_DOMAIN    NULL
> > #   define STUN_SERVER    NULL
> > #endif
> >
> > //
> > // Use ICE?
> > //
> > #define USE_ICE        1
>
> I use asterisk which has this IP 192.168.1.73, and calling to user 'echo',
> asterisk answeres and returns every word you said in your microfone to your
> speaker. I think that over the console of the E65 runing symbian_ua it just
> need to press 'm' (number 6) to call sip:echo at 192.168.1.73, please no doubt
> to tell me what you think, any hint could help me.
>
> thanks.
>
>
> On Feb 13, 2008 1:21 PM, Nanang Izzuddin <nanang.izzuddin at gmail.com> wrote:
> > Hi Rodrigo,
> >
> > I did couples calls using pjsip on Symbian (and it was E65 as well
> > :D), directly or via registrar, from/to pjsua PC.
> > But I have never experienced the assertion.
> >
> > Perhaps you need to investigate more detail on the call stack trace,
> > in case there is something strange.
> >
> > Regards,
> > nanang
> >
> >
> >
> >
> >
> > On 12/02/2008, Rodrigo Vega <vegaroy13 at gmail.com> wrote:
> > > Hi benny:
> > >
> > > First of all I want to thank you for all your support.
> > >
> > > Finaly I can make calls from pjsua_wince over an ipaq hp. It was
> problems of
> > > installations, but solved successfully.
> > >
> > > Now I'm working over a phone Nokia E65 which it's compatible with the
> > > Building and Debugging PJSIP on Symbian S60 3rd Edition Device using
> Carbide
> > > C++ tutorial.
> > >
> > > Compilations works.
> > > Sending the application to the phone works.
> > > It runs.
> > >
> > > I'm doing a call from the ipaq hp to the symbian phone.
> > >
> > > Asterisk says this:
> > >
> > > [Feb 12 15:56:49] NOTICE[10306]: chan_sip.c:12517
> handle_response_peerpoke:
> > > Peer 'symbian' is now Reachable. (1211ms / 2000ms)
> > >  [Feb 12 15:56:52] NOTICE[10306]: chan_sip.c:12517
> handle_response_peerpoke:
> > > Peer 'wincewm' is now Reachable. (1024ms / 2000ms)
> > >     -- Executing [711 at internal:1] Dial("SIP/wincewm-081ed998",
> > > "SIP/symbian") in new stack
> > >      -- Called symbian
> > >     -- SIP/symbian-081f78d0 is ringing
> > >     -- SIP/symbian-081f78d0 answered SIP/wincewm-081ed998
> > >     -- Packet2Packet bridging SIP/wincewm-081ed998 and
> SIP/symbian-081f78d0
> > > [Feb 12 15:57:53] NOTICE[10306]: chan_sip.c:15655 sip_poke_noanswer:
> Peer
> > > 'symbian' is now UNREACHABLE!  Last qualify: 1211
> > >
> > > I press the button 1 on the symbian phone (like pressing buttom 'a' of
> > > answere).
> > >
> > > I cannot read every thins that the console prints on the phone, I can
> see
> > > something about RTP... finaly shows up this text:
> > >
> > > assertion "!" Unsupported address family"" failed: file
> > > "..\\pjlib\\src\\/os_symbian.h", line 295
> > >
> > > and then ask to press any key, and the application dies.
> > >
> > > My macro's config are these ones:
> > >
> > > //
> > > // Basic config.
> > > //
> > > #define SIP_PORT    5060
> > >
> > >
> > > //
> > > // Destination URI (to make call, or to subscribe presence)
> > > //
> > > #define SIP_DST_URI    "sip:echo at 192.168.1.73"
> > >
> > > //
> > > // Account
> > > //
> > > #define HAS_SIP_ACCOUNT    1    // 0 to disable registration
> > > #define SIP_DOMAIN    "192.168.1.73"
> > > #define SIP_USER    "symbian"
> > >  #define SIP_PASSWD    "symb"
> > >
> > > //
> > > // Outbound proxy for all accounts
> > > //
> > > #define SIP_PROXY    NULL
> > > #define SIP_PROXY    "sip:192.168.1.73:lr"
> > >
> > >
> > > //
> > > // Configure nameserver if DNS SRV is to be used with both SIP
> > >  // or STUN (for STUN see other settings below)
> > > //
> > > #define NAMESERVER    NULL
> > > //#define NAMESERVER    "192.168.0.1"
> > >
> > > //
> > > // STUN server
> > > #if 0
> > >     // Use this to have the STUN server resolved normally
> > >  #   define STUN_DOMAIN    NULL
> > > #   define STUN_SERVER    "stun.xten.com"
> > > #elif 0
> > >     // Use this to have the STUN server resolved with DNS SRV
> > > #   define STUN_DOMAIN    "iptel.org"
> > >  #   define STUN_SERVER    NULL
> > > #else
> > >     // Use this to disable STUN
> > > #   define STUN_DOMAIN    NULL
> > > #   define STUN_SERVER    NULL
> > > #endif
> > >
> > > //
> > > // Use ICE?
> > > //
> > > #define USE_ICE        1
> > >
> > >
> > >
> > > I also had another problem which was solved doing this:
> > >
> > > cfg.cred_info[0].realm =
> pj_str("*");//pj_str(SIP_DOMAIN);
> > >
> > > because for asterisk could be "asterisk" or "*" that is the wild-card.
> > >
> > >
> > > Which is the problem here?
> > >
> > > Thanks for your support.
> > >
> > > _______________________________________________
> > >  Visit our blog: http://blog.pjsip.org
> > >
> > >  pjsip mailing list
> > >  pjsip at lists.pjsip.org
> > >
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> > >
> > >
> >
> > _______________________________________________
> > Visit our blog: http://blog.pjsip.org
> >
> > pjsip mailing list
> > pjsip at lists.pjsip.org
> >
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
>
>
> _______________________________________________
>  Visit our blog: http://blog.pjsip.org
>
>  pjsip mailing list
>  pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>



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