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merged jcmoore's patch for configurable SDP origin-field username and session field, closes issue# 10795 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Dwayne M. Hubbard 2007-09-24 17:10:14 +00:00
parent 2d95565f53
commit 0f53904918
2 changed files with 28 additions and 4 deletions
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27
channels/chan_sip.c
View File

@ -553,6 +553,8 @@ static const struct cfsip_options {
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#endif
/* Default setttings are used as a channel setting and as a default when
@ -614,6 +616,8 @@ static int dumphistory; /*!< Dump history to verbose before destroying SIP dia
static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
static int allow_external_domains; /*!< Accept calls to external SIP domains? */
static int global_callevents; /*!< Whether we send manager events or not */
static int global_t1min; /*!< T1 roundtrip time minimum */
@ -6947,8 +6951,8 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
p->t38.jointcapability);
}
ast_str_append(&m_modem, 0, "v=0\r\n");
ast_str_append(&m_modem, 0, "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "s=session\r\n");
ast_str_append(&m_modem, 0, "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner , p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
ast_str_append(&m_modem, 0, "c=IN IP4 %s\r\n", ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "t=0 0\r\n");
ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));
@ -7054,7 +7058,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
/* SDP fields */
char *version = "v=0\r\n"; /* Protocol version */
char *subject = "s=session\r\n"; /* Subject of the session */
char subject[256]; /* Subject of the session */
char owner[256]; /* Session owner/creator */
char connection[256]; /* Connection data */
char *stime = "t=0 0\r\n"; /* Time the session is active */
@ -7079,6 +7083,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
char codecbuf[BUFSIZ];
char buf[BUFSIZ];
/* Set the SDP session name */
snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return AST_FAILURE;
@ -7172,7 +7179,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
@ -11642,6 +11649,8 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " Call limit peers only: %s\n", cli_yesno(global_limitonpeers));
ast_cli(a->fd, " Direct RTP setup: %s\n", cli_yesno(global_directrtpsetup));
ast_cli(a->fd, " User Agent: %s\n", global_useragent);
ast_cli(a->fd, " SDP Session Name: %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
ast_cli(a->fd, " SDP Owner Name: %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
ast_cli(a->fd, " Reg. context: %s\n", S_OR(global_regcontext, "(not set)"));
ast_cli(a->fd, " Regexten on Qualify: %s\n", cli_yesno(global_regextenonqualify));
ast_cli(a->fd, " Caller ID: %s\n", default_callerid);
@ -17858,6 +17867,8 @@ static int reload_config(enum channelreloadreason reason)
global_alwaysauthreject = 0;
global_allowsubscribe = FALSE;
snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ASTERISK_VERSION);
snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ASTERISK_VERSION);
snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(global_realm));
ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
@ -17930,6 +17941,14 @@ static int reload_config(enum channelreloadreason reason)
} else if (!strcasecmp(v->name, "useragent")) {
ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
} else if (!strcasecmp(v->name, "sdpsession")) {
ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
} else if (!strcasecmp(v->name, "sdpowner")) {
/* Field cannot contain spaces */
if (!strstr(v->value, " "))
ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
else
ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d. Using default.\n", v->value, v->lineno);
} else if (!strcasecmp(v->name, "allowtransfer")) {
global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
} else if (!strcasecmp(v->name, "rtcachefriends")) {

5
configs/sip.conf.sample
View File

@ -113,6 +113,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable