On 27.07.20 19:08, Andreas Wehrmann wrote:
> Am 27.07.2020 um 16:53 schrieb r.b.muller@xxxxxxxxx:
>>
>> @ Some Dick
>>
>> Thanks for taking the time to reply.
>>
>> If this is done at registration, I assume this is where the magic
>> should happen, in the account configuration
>>
>> https://www.pjsip.org/pjsip/docs/html/structpjsua__acc__config.htm#a08473de6401e966d23f34d3a9a05bdd0
>> <https://www.pjsip.org/pjsip/docs/html/structpjsua__acc__config.htm#a08473de6401e966d23f34d3a9a05bdd0>
>>
>>
>> I am unclear as what element of the account struct to set to the
>> outgoing spoofed number.
>>
>> Can you guide me?
>>
>> Met Vriendelijke Groet, Regards,
>>
>> Rob Muller
>>
>> mailto:r.b.muller@xxxxxxxxx
>>
> I actually implemented setting a custom "From" user and display name
> for pjsua_call_make_call();
> however, I had to patch PJ for this.
> Not sure right now if I've pushed this to my public repo, I'll check
> tomorrow.
> If not, I'll just post the changes here on the mailing list.
>
> Best regards,
>
> Andreas
Okay, not in my repo.
I modified pjsua_call_make_call() and the pjsua_call_setting struct in
pjsua.h.
See the attachement for my versions.
All the best,
Andreas
/**
* Call settings.
*/
typedef struct pjsua_call_setting
{
/**
* Bitmask of #pjsua_call_flag constants.
*
* Default: PJSUA_CALL_INCLUDE_DISABLED_MEDIA
*/
unsigned flag;
/**
* This flag controls what methods to request keyframe are allowed on
* the call. Value is bitmask of #pjsua_vid_req_keyframe_method.
*
* Default: (PJSUA_VID_REQ_KEYFRAME_SIP_INFO |
* PJSUA_VID_REQ_KEYFRAME_RTCP_PLI)
*/
unsigned req_keyframe_method;
/**
* Number of simultaneous active audio streams for this call. Setting
* this to zero will disable audio in this call.
*
* Default: 1
*/
unsigned aud_cnt;
/**
* Number of simultaneous active video streams for this call. Setting
* this to zero will disable video in this call.
*
* Default: 1 (if video feature is enabled, otherwise it is zero)
*/
unsigned vid_cnt;
/**
* When initially making a call, this can be used to enable the setting
* of a custom display name in the From header of the outgoing INVITE.
*
* Default: Use display name from account id (if any)
*/
pj_bool_t set_from_display_name;
/**
* When use_from_display_name is enabled, the string here will be set
* as the display name in the From header of the outgoing INVITE.
*
* Default: Use display name from account id (if any)
*/
pj_str_t from_display_name;
/**
* When initially making a call, this can be used to set the user part
* of the From header in the outgoing INVITE.
*
* Default: Use account id
*/
pj_str_t from_user;
} pjsua_call_setting;
/*
* Make outgoing call to the specified URI using the specified account.
*/
PJ_DEF(pj_status_t) pjsua_call_make_call(pjsua_acc_id acc_id,
const pj_str_t *dest_uri,
const pjsua_call_setting *opt,
void *user_data,
const pjsua_msg_data *msg_data,
pjsua_call_id *p_call_id)
{
pj_pool_t *tmp_pool = NULL;
pjsip_dialog *dlg = NULL;
pjsua_acc *acc;
pjsua_call *call;
int call_id = -1;
pj_str_t contact;
pj_status_t status;
pjsip_sip_uri* from_uri;
pjsip_name_addr* from_na;
pj_str_t from_header_str;
pj_str_t custom_from_user;
pj_str_t custom_from_display_name;
char new_from_uri[ PJSIP_MAX_URL_SIZE ];
int len;
/* Check that account is valid */
PJ_ASSERT_RETURN(acc_id>=0 || acc_id<(int)PJ_ARRAY_SIZE(pjsua_var.acc),
PJ_EINVAL);
/* Check arguments */
PJ_ASSERT_RETURN(dest_uri, PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Making call with acc #%d to %.*s", acc_id,
(int)dest_uri->slen, dest_uri->ptr));
pj_log_push_indent();
PJSUA_LOCK();
acc = &pjsua_var.acc[acc_id];
if (!acc->valid) {
pjsua_perror(THIS_FILE, "Unable to make call because account "
"is not valid", PJ_EINVALIDOP);
status = PJ_EINVALIDOP;
goto on_error;
}
/* Find free call slot. */
call_id = alloc_call_id();
if (call_id == PJSUA_INVALID_ID) {
pjsua_perror(THIS_FILE, "Error making call", PJ_ETOOMANY);
status = PJ_ETOOMANY;
goto on_error;
}
/* Clear call descriptor */
reset_call(call_id);
call = &pjsua_var.calls[call_id];
/* Associate session with account */
call->acc_id = acc_id;
call->call_hold_type = acc->cfg.call_hold_type;
/* Generate per-session RTCP CNAME, according to RFC 7022. */
pj_create_random_string(call->cname_buf, call->cname.slen);
/* Apply call setting */
status = apply_call_setting(call, opt, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
goto on_error;
}
/* Create sound port if none is instantiated, to check if sound device
* can be used. But only do this with the conference bridge, as with
* audio switchboard (i.e. APS-Direct), we can only open the sound
* device once the correct format has been known
*/
if (!pjsua_var.is_mswitch && pjsua_var.snd_port==NULL &&
pjsua_var.null_snd==NULL && !pjsua_var.no_snd && call->opt.aud_cnt > 0)
{
status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev);
if (status != PJ_SUCCESS)
goto on_error;
}
/* Create temporary pool */
tmp_pool = pjsua_pool_create("tmpcall10", 512, 256);
/* Verify that destination URI is valid before calling
* pjsua_acc_create_uac_contact, or otherwise there
* a misleading "Invalid Contact URI" error will be printed
* when pjsua_acc_create_uac_contact() fails.
*/
if (1) {
pjsip_uri *uri;
pj_str_t dup;
pj_strdup_with_null(tmp_pool, &dup, dest_uri);
uri = pjsip_parse_uri(tmp_pool, dup.ptr, dup.slen, 0);
if (uri == NULL) {
pjsua_perror(THIS_FILE, "Unable to make call",
PJSIP_EINVALIDREQURI);
status = PJSIP_EINVALIDREQURI;
goto on_error;
}
}
/* Mark call start time. */
pj_gettimeofday(&call->start_time);
/* Reset first response time */
call->res_time.sec = 0;
/* Create suitable Contact header unless a Contact header has been
* set in the account.
*/
if (acc->contact.slen) {
contact = acc->contact;
} else {
status = pjsua_acc_create_uac_contact(tmp_pool, &contact,
acc_id, dest_uri);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to generate Contact header",
status);
goto on_error;
}
}
/* Build From header (if any)
*/
pj_strassign(&from_header_str, &acc->cfg.id);
if(opt) {
if( (pj_strlen(&opt->from_user) > 0) || opt->set_from_display_name ) {
pj_strdup_with_null(tmp_pool, &from_header_str, &acc->cfg.id);
from_na = (pjsip_name_addr*)pjsip_parse_uri(tmp_pool,
(char*)pj_strbuf( &from_header_str ),
pj_strlen( &from_header_str ),
PJSIP_PARSE_URI_AS_NAMEADDR );
if (! from_na) {
status = PJSIP_EINVALIDURI;
pjsua_perror(THIS_FILE, "Failed to parse account ID",
status);
goto on_error;
}
from_uri = pjsip_uri_get_uri(from_na);
if(! from_uri) {
status = PJSIP_EINVALIDURI;
pjsua_perror(THIS_FILE, "Failed to get account URI",
status);
goto on_error;
}
if( pj_strlen(&opt->from_user) > 0 ) {
pj_strassign( &from_uri->user,
(pj_str_t*)&opt->from_user );
}
if( opt->set_from_display_name ) {
pj_strassign( &from_na->display,
(pj_str_t*)&opt->from_display_name );
}
/* From name-addr has been updated, print the entire thing now
*/
len = pjsip_uri_print( PJSIP_URI_IN_FROMTO_HDR, from_na, new_from_uri, PJSIP_MAX_URL_SIZE );
if( len < 0 ) {
status = PJSIP_EINVALIDREQURI;
pjsua_perror(THIS_FILE, "Failed to print new From URI",
status);
goto on_error;
}
pj_strset( &from_header_str,
new_from_uri, len );
}
}
/* Create outgoing dialog: */
status = pjsip_dlg_create_uac( pjsip_ua_instance(),
&from_header_str, &contact,
dest_uri,
(msg_data && msg_data->target_uri.slen?
&msg_data->target_uri: dest_uri),
&dlg);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Dialog creation failed", status);
goto on_error;
}
/* Increment the dialog's lock otherwise when invite session creation
* fails the dialog will be destroyed prematurely.
*/
pjsip_dlg_inc_lock(dlg);
dlg_set_via(dlg, acc);
/* Calculate call's secure level */
call->secure_level = get_secure_level(acc_id, dest_uri);
/* Attach user data */
call->user_data = user_data;
/* Store variables required for the callback after the async
* media transport creation is completed.
*/
if (msg_data) {
call->async_call.call_var.out_call.msg_data = pjsua_msg_data_clone(
dlg->pool, msg_data);
}
call->async_call.dlg = dlg;
/* Temporarily increment dialog session. Without this, dialog will be
* prematurely destroyed if dec_lock() is called on the dialog before
* the invite session is created.
*/
pjsip_dlg_inc_session(dlg, &pjsua_var.mod);
if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) {
/* Init media channel */
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC,
call->secure_level, dlg->pool,
NULL, NULL, PJ_TRUE,
&on_make_call_med_tp_complete);
}
if (status == PJ_SUCCESS) {
status = on_make_call_med_tp_complete(call->index, NULL);
if (status != PJ_SUCCESS)
goto on_error;
} else if (status != PJ_EPENDING) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
goto on_error;
}
/* Done. */
if (p_call_id)
*p_call_id = call_id;
pjsip_dlg_dec_lock(dlg);
pj_pool_release(tmp_pool);
PJSUA_UNLOCK();
pj_log_pop_indent();
return PJ_SUCCESS;
on_error:
if (dlg) {
/* This may destroy the dialog */
pjsip_dlg_dec_lock(dlg);
}
if (call_id != -1) {
pjsua_media_channel_deinit(call_id);
reset_call(call_id);
}
pjsua_check_snd_dev_idle();
if (tmp_pool)
pj_pool_release(tmp_pool);
PJSUA_UNLOCK();
pj_log_pop_indent();
return status;
}
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
