@ -1053,6 +1053,7 @@ struct sip_pvt {
AST_STRING_FIELD ( fromuser ) ; /*!< User to show in the user field */
AST_STRING_FIELD ( fromname ) ; /*!< Name to show in the user field */
AST_STRING_FIELD ( tohost ) ; /*!< Host we should put in the "to" field */
AST_STRING_FIELD ( todnid ) ; /*!< DNID of this call (overrides host) */
AST_STRING_FIELD ( language ) ; /*!< Default language for this call */
AST_STRING_FIELD ( mohinterpret ) ; /*!< MOH class to use when put on hold */
AST_STRING_FIELD ( mohsuggest ) ; /*!< MOH class to suggest when putting a peer on hold */
@ -7839,17 +7840,30 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
if ( p - > options & & p - > options - > uri_options )
ast_str_append ( & invite , 0 , " ;%s " , p - > options - > uri_options ) ;
/* This is the request URI, which is the next hop of the call
which may or may not be the destination of the call
*/
ast_string_field_set ( p , uri , invite - > str ) ;
if ( ! ast_strlen_zero ( p - > todnid ) ) {
/*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
if ( ! strchr ( p - > todnid , ' @ ' ) ) {
/* We have no domain in the dnid */
snprintf ( to , sizeof ( to ) , " <sip:%s@%s>%s%s " , p - > todnid , p - > tohost , ast_strlen_zero ( p - > theirtag ) ? " " : " ;tag= " , p - > theirtag ) ;
} else {
snprintf ( to , sizeof ( to ) , " <sip:%s>%s%s " , p - > todnid , ast_strlen_zero ( p - > theirtag ) ? " " : " ;tag= " , p - > theirtag ) ;
}
} else {
if ( sipmethod = = SIP_NOTIFY & & ! ast_strlen_zero ( p - > theirtag ) ) {
/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
snprintf ( to , sizeof ( to ) , " <%s%s>;tag=%s " , ( strncasecmp ( p - > uri , " sip: " , 4 ) ? " " : " sip: " ) , p - > uri , p - > theirtag ) ;
} else if ( p - > options & & p - > options - > vxml_url ) {
/* If there is a VXML URL append it to the SIP URL */
snprintf ( to , sizeof ( to ) , " <%s>;%s " , p - > uri , p - > options - > vxml_url ) ;
} else
snprintf ( to , sizeof ( to ) , " <%s> " , p - > uri ) ;
}
if ( sipmethod = = SIP_NOTIFY & & ! ast_strlen_zero ( p - > theirtag ) ) {
/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
snprintf ( to , sizeof ( to ) , " <%s%s>;tag=%s " , ( strncasecmp ( p - > uri , " sip: " , 4 ) ? " " : " sip: " ) , p - > uri , p - > theirtag ) ;
} else if ( p - > options & & p - > options - > vxml_url ) {
/* If there is a VXML URL append it to the SIP URL */
snprintf ( to , sizeof ( to ) , " <%s>;%s " , p - > uri , p - > options - > vxml_url ) ;
} else
snprintf ( to , sizeof ( to ) , " <%s> " , p - > uri ) ;
init_req ( req , sipmethod , p - > uri ) ;
/* now tmp_n is available so reuse it to build the CSeq */
snprintf ( tmp_n , sizeof ( tmp_n ) , " %d %s " , + + p - > ocseq , sip_methods [ sipmethod ] . text ) ;
@ -7858,6 +7872,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
add_header ( req , " Max-Forwards " , DEFAULT_MAX_FORWARDS ) ;
/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
* OTOH , then we won ' t have anything in p - > route anyway */
/* Build Remote Party-ID and From */
if ( ast_test_flag ( & p - > flags [ 0 ] , SIP_SENDRPID ) & & ( sipmethod = = SIP_INVITE ) ) {
build_rpid ( p ) ;
@ -17304,7 +17319,13 @@ static int sip_devicestate(void *data)
}
/*! \brief PBX interface function -build SIP pvt structure
SIP calls initiated by the PBX arrive here */
SIP calls initiated by the PBX arrive here
SIP Dial string syntax
SIP / exten @ host ! dnid
or SIP / host / exten ! dnid
or SIP / host ! dnid
*/
static struct ast_channel * sip_request_call ( const char * type , int format , void * data , int * cause )
{
struct sip_pvt * p ;
@ -17312,6 +17333,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
char * ext , * host ;
char tmp [ 256 ] ;
char * dest = data ;
char * dnid ;
int oldformat = format ;
/* mask request with some set of allowed formats.
@ -17344,7 +17366,18 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
return NULL ;
}
/* Save the destination, the SIP dial string */
ast_copy_string ( tmp , dest , sizeof ( tmp ) ) ;
/* Find DNID and take it away */
dnid = strchr ( tmp , ' ! ' ) ;
if ( dnid ! = NULL ) {
* dnid + + = ' \0 ' ;
ast_string_field_set ( p , todnid , dnid ) ;
}
/* Find at sign - @ */
host = strchr ( tmp , ' @ ' ) ;
if ( host ) {
* host + + = ' \0 ' ;
@ -17356,6 +17389,11 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
host = tmp ;
}
/* We now have
host = peer name , DNS host name or DNS domain ( for SRV )
ext = extension ( user part of URI )
dnid = destination of the call ( applies to the To : header )
*/
if ( create_addr ( p , host ) ) {
* cause = AST_CAUSE_UNREGISTERED ;
ast_debug ( 3 , " Cant create SIP call - target device not registred \n " ) ;
@ -17372,7 +17410,8 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
/* We have an extension to call, don't use the full contact here */
/* This to enable dialing registered peers with extension dialling,
like SIP / peername / extension
SIP / peername will still use the full contact */
SIP / peername will still use the full contact
*/
if ( ext ) {
ast_string_field_set ( p , username , ext ) ;
ast_string_field_set ( p , fullcontact , NULL ) ;
Olle Johansson
18 years ago