The get_pai function in chan_sip.c didn't recognized a proper callerid name and

number from a P-Asserted-Identity cause the header parsing logic was wrong. 
Changing the parsing functions to the sip header parsing APIs in 
reqresp_parser.h solves this problem.

Review: https://reviewboard.asterisk.org/r/1673
Reviewed by: wdoekes2 and Mark Michelson
........

Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@351408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
Stefan Schmidt 14 years ago
parent e80559676f
commit 57b83daf59

@ -15207,10 +15207,12 @@ static int get_pai(struct sip_pvt *p, struct sip_request *req)
{
char pai[256];
char privacy[64];
char *cid_num = "";
char *cid_name = "";
char *cid_num = NULL;
char *cid_name = NULL;
char emptyname[1] = "";
int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
char *start = NULL, *end = NULL, *uri = NULL;
char *uri = NULL;
int is_anonymous = 0, do_update = 1, no_name = 0;
ast_copy_string(pai, sip_get_header(req, "P-Asserted-Identity"), sizeof(pai));
@ -15218,73 +15220,59 @@ static int get_pai(struct sip_pvt *p, struct sip_request *req)
return 0;
}
start = pai;
if (*start == '"') {
*start++ = '\0';
end = strchr(start, '"');
if (!end) {
return 0;
}
*end++ = '\0';
cid_name = start;
start = ast_skip_blanks(end);
/* use the reqresp_parser function get_name_and_number*/
if (get_name_and_number(pai, &cid_name, &cid_num)) {
return 0;
}
if (*start != '<')
return 0;
/* At this point, 'start' points to the URI in brackets.
* We need a copy so that our comparison to the anonymous
* URI is valid.
*/
uri = ast_strdupa(start);
*start++ = '\0';
end = strchr(start, '@');
if (!end) {
return 0;
if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num)) {
ast_shrink_phone_number(cid_num);
}
*end++ = '\0';
uri = get_in_brackets(pai);
if (!strncasecmp(uri, "sip:anonymous@anonymous.invalid", 31)) {
callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
/*XXX Assume no change in cid_num. Perhaps it should be
* blanked?
*/
ast_free(cid_num);
is_anonymous = 1;
cid_num = (char *)p->cid_num;
} else if (!strncasecmp(start, "sip:", 4)) {
cid_num = start + 4;
if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num))
ast_shrink_phone_number(cid_num);
start = end;
end = strchr(start, '>');
if (!end) {
return 0;
}
*end = '\0';
} else {
return 0;
}
ast_copy_string(privacy, sip_get_header(req, "Privacy"), sizeof(privacy));
if (!ast_strlen_zero(privacy) && strncmp(privacy, "id", 2)) {
callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
}
if (!cid_name) {
no_name = 1;
cid_name = (char *)emptyname;
}
/* Only return true if the supplied caller id is different */
if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres) {
return 0;
}
do_update = 0;
} else {
ast_string_field_set(p, cid_num, cid_num);
ast_string_field_set(p, cid_name, cid_name);
p->callingpres = callingpres;
ast_string_field_set(p, cid_num, cid_num);
ast_string_field_set(p, cid_name, cid_name);
p->callingpres = callingpres;
if (p->owner) {
ast_set_callerid(p->owner, cid_num, cid_name, NULL);
p->owner->caller.id.name.presentation = callingpres;
p->owner->caller.id.number.presentation = callingpres;
if (p->owner) {
ast_set_callerid(p->owner, cid_num, cid_name, NULL);
p->owner->caller.id.name.presentation = callingpres;
p->owner->caller.id.number.presentation = callingpres;
}
}
return 1;
/* get_name_and_number allocates memory for cid_num and cid_name so we have to free it */
if (!is_anonymous) {
ast_free(cid_num);
}
if (!no_name) {
ast_free(cid_name);
}
return do_update;
}
/*! \brief Get name, number and presentation from remote party id header,

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