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.
>
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