Oh yes, I did try the register too, thanks.
I need to clarify few things coz all I have is the source
code(pjproject-1.3) and my laptop to work with it. But I had to make sure
that this script works fine, if somebody else is gonna use to make a call (
with proper sip provider )
Guess with source compiled properly, I can use DOS prompt to run my
executable and call.
I had another script written where I included the fn pjsua_invite(buf,
&inv), but when I compile I get error: undefined reference pjsua_invite.
Thought I'm missing some library, so included the pjsua_invite.c file in
C:\pjproject-1.3\pjsip\src\pjsua-lib.
and also included this below part in:
(C:\pjproject-1.3\pjsip\include\pjsua-lib)
** PJSUA instance. */
extern struct pjsua pjsua;
/*****************************************************************************
* PJSUA Invite session API (defined in pjsua_inv.c).
*/
/**
* Make outgoing call.
*/
pj_status_t pjsua_invite(const char *cstr_dest_uri,
pjsip_inv_session **p_inv);
/**
* Handle incoming invite request.
*/
pj_bool_t pjsua_inv_on_incoming(pjsip_rx_data *rdata);
/**
* Callback to be called by session when invite session's state has changed.
*/
void pjsua_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e);
/**
* Callback to be called by session when outgoing dialog has forked.
* This function will create a forked dialog.
*/
void pjsua_inv_on_new_session(pjsip_inv_session *inv, pjsip_event *e);
/**
* Callback to be called when SDP offer/answer negotiation has just
completed
* in the session. This function will start/update media if negotiation
* has succeeded.
*/
void pjsua_inv_on_media_update(pjsip_inv_session *inv, pj_status_t status);
/*****************************************************************************
* User Interface API.
*
* The UI API specifies functions that will be called by pjsua upon
* occurence of various events.
*/
/**
* Notify UI when invite state has changed.
*/
void pjsua_ui_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e);
But still I'm getting the same error. Not sure what I'm missing here?
Thanks!
2009/9/11 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>
> pjsua_simple.c is self documented:
>
> /**
> * simple_pjsua.c
> *
> * This is a very simple but fully featured SIP user agent, with the
> * following capabilities:
> * - SIP registration
> * - Making and receiving call
> * - Audio/media to sound device.
> *
> * Usage:
> * - To make outgoing call, start simple_pjsua with the URL of remote
> * destination to contact.
> * E.g.:
> * simpleua sip:user at remote
> *
> * - Incoming calls will automatically be answered with 200.
> *
> * This program will quit once it has completed a single call.
> */
>
> Also, if you want simple_pjsua to register, you'll need to change this
> values and recompile it:
>
> #define SIP_DOMAIN "example.com"
> #define SIP_USER "alice"
> #define SIP_PASSWD "secret"
>
> Seems like the sample itself is counting on translating "example.com" as
> the DNS to your registrar/proxy.
>
> And yes, you're right about the 'h' key.
>
>
> 2009/9/11 Rachel Baskaran <rachelbaskaran at gmail.com>
>
>> Thanks that did help!
>>
>> But when I press 'h' it says, APP Call 0 state=DISCONNECTED. so this means
>> I talk to the user (sip:user at example.com <sip%3Auser at example.com> ) and
>> when I press 'h' it terminates the call to the user I dialed right?
>>
>> Will I able to do the same if I place a phone number (
>> 418-345-6789 at sipprovider.com )?
>> Provided I have a proper dial plan and SIP PROXY to handle the call.
>>
>> Or do I need to write any separate script to do the dialing to a number?
>>
>>
>> Thanks!
>>
>> 2009/9/11 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>
>>
>>> perhaps the fatal exception comes from the uninitialized uri variable.
>>>
>>> Anyway, this is how I found that line in the source:
>>> if (argc > 1) {
>>> pj_str_t uri = pj_str(argv[1]);
>>> status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
>>> if (status != PJ_SUCCESS) error_exit("Error making call", status);
>>> }
>>>
>>>
>>> 2009/9/11 Rachel Baskaran <rachelbaskaran at gmail.com>
>>>
>>>> Alright, it showed me error undeclared in URI in below line
>>>> pj_str_t uri=pj_str(URI);
>>>> so I changed to
>>>> pj_str_t uri;
>>>>
>>>> And entered my exe folder(pjsip-apps/bin/rach/i686-pc-mingw32)
>>>> simple_pjsua sip:rachel at example.com <sip%3Arachel at example.com> ->
>>>> Enter
>>>>
>>>> It registered but a fatal error was displayed:
>>>> pjsua_acc.c Registration sent
>>>> pjsua_media.c Opening sound device
>>>> except.c !!! FATAL: unhandled exception PJLIB/no memory
>>>> Assertion failed: handler != ((void *)0), file ...src/pj/except.c, line
>>>> 50.
>>>>
>>>>
>>>> If run the script like simple_pjsua.c ->Enter
>>>>
>>>> It runs the header for few mins and displays, SIP registration failed
>>>> timeout. Since I'm testing just typed sip:user at example.com<sip%3Auser at example.com>-> Enter, it just displayed the options 'h' hangup the calls and 'q' to
>>>> quit.
>>>>
>>>> Am I building it in right way, if YES, Why do I get that FATAL error?
>>>>
>>>> Thanks!
>>>>
>>>>
>>>> 2009/9/11 Paulo Rog?rio Panhoto <paulo at voicetechnology.com.br>
>>>>
>>>> argv[1] is the first argument you pass to a program in command line.
>>>>> i.e. simple_ua.exe sip:user at example.org <sip%3Auser at example.org>
>>>>>
>>>>> 2009/9/11 Rachel Baskaran <rachelbaskaran at gmail.com>
>>>>>
>>>>>> Hey,
>>>>>>
>>>>>> I'm trying to edit the simple_pjsua.c to call a specified URL using
>>>>>> pjsua_call_make_call. The proram compiled successfully and registers too,
>>>>>> but doesn't call the URL since I didn't know where or how to define my
>>>>>> destination URL in main fn()
>>>>>>
>>>>>> int main(int argc,char *argv[])
>>>>>> {
>>>>>> pjsua_acc_id acc_id;
>>>>>> pj_status_t status;
>>>>>>
>>>>>>
>>>>>>
>>>>>> status = pjsua_create();
>>>>>> if(status != PJ_SUCCESS) error_exit("Error in pjsua_create()",
>>>>>> status);
>>>>>>
>>>>>>
>>>>>> if(argc > 1)
>>>>>> {
>>>>>> status = pjsua_verify_sip_url(argv[1]);
>>>>>> if(status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);
>>>>>> }
>>>>>>
>>>>>> /* Init pjsua */
>>>>>> {
>>>>>>
>>>>>> pjsua_config cfg;
>>>>>> pjsua_logging_config log_cfg;
>>>>>>
>>>>>>
>>>>>> pjsua_config_default(&cfg);
>>>>>> cfg.cb.on_incoming_call = &on_incoming_call;
>>>>>> cfg.cb.on_call_media_state = &on_call_media_state;
>>>>>> cfg.cb.on_call_state = &on_call_state;
>>>>>>
>>>>>>
>>>>>>
>>>>>> pjsua_logging_config_default(&log_cfg);
>>>>>> log_cfg.console_level = 4;
>>>>>>
>>>>>>
>>>>>> status = pjsua_init(&cfg, &log_cfg, NULL);
>>>>>> if(status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
>>>>>> }
>>>>>>
>>>>>>
>>>>>>
>>>>>> /* add UDP transport */
>>>>>> {
>>>>>>
>>>>>> pjsua_transport_config cfg;
>>>>>> pjsua_transport_config_default(&cfg);
>>>>>> cfg.port = 5060;
>>>>>> status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
>>>>>> if(status != PJ_SUCCESS) error_exit("Error creating transport",
>>>>>> status);
>>>>>> }
>>>>>> /* start pjsua */
>>>>>>
>>>>>> status = pjsua_start();
>>>>>> if(status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
>>>>>>
>>>>>> /* Register to SIP server creating an account */
>>>>>>
>>>>>> {
>>>>>> pjsua_acc_config cfg;
>>>>>>
>>>>>>
>>>>>> pjsua_acc_config_default(&cfg);
>>>>>> cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
>>>>>> cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
>>>>>> cfg.cred_count = 1;
>>>>>> cfg.cred_info[0].realm = pj_str(SIP_DOMAIN);
>>>>>> cfg.cred_info[0].scheme = pj_str("digest");
>>>>>> cfg.cred_info[0].username = pj_str(SIP_USER);
>>>>>> cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
>>>>>> cfg.cred_info[0].data = pj_str(SIP_PASSWD);
>>>>>>
>>>>>> status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
>>>>>> if(status != PJ_SUCCESS) error_exit("Error adding account", status);
>>>>>> }
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> if(argc > 1)
>>>>>> {
>>>>>>
>>>>>> pj_str_t uri=pj_str(URI);
>>>>>>
>>>>>> status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
>>>>>> if(status != PJ_SUCCESS) error_exit("Error making call", status);
>>>>>> }
>>>>>>
>>>>>> Any help would be great!
>>>>>>
>>>>>> Rachel
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>
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