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Add user=phone option (bug #2244, thanks oej)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Mark Spencer 21 years ago
parent f375ebee86
commit 1a147cf804
2 changed files with 48 additions and 5 deletions
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50
channels/chan_sip.c
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@ -42,6 +42,7 @@
#include <asterisk/astdb.h>
#include <asterisk/causes.h>
#include <asterisk/utils.h>
#include <asterisk/file.h>
#ifdef OSP_SUPPORT
#include <asterisk/astosp.h>
#endif
@ -197,8 +198,9 @@ static int videosupport = 0;
static int compactheaders = 0; /* send compact sip headers */
static int global_dtmfmode = SIP_DTMF_RFC2833; /* DTMF mode default */
static int recordhistory = 0;
static int recordhistory = 0; /* Record SIP history. Off by default */
static int global_promiscredir; /* Support of 302 REDIR - Default off */
static int global_usereqphone; /* User=phone support, default 0 */
static char global_musicclass[MAX_LANGUAGE] = ""; /* Global music on hold class */
static char global_realm[AST_MAX_EXTENSION] = "asterisk"; /* Default realm */
@ -346,6 +348,7 @@ static struct sip_pvt {
int stateid;
int dialogver;
int promiscredir; /* Promiscuous redirection */
int usereqphone; /* Add user=phone to numeric URI. Default off */
int trustrpid; /* Trust RPID headers? */
int progressinband;
@ -458,6 +461,7 @@ struct sip_peer {
int trustrpid; /* Trust Remote Party ID headers? */
int useclientcode; /* SNOM clientcode support */
int progressinband;
int usereqphone; /* Add user=phone to URI. Default off */
struct sockaddr_in addr; /* IP address of peer */
struct in_addr mask;
@ -1292,6 +1296,7 @@ static int create_addr(struct sip_pvt *r, char *opeer)
r->noncodeccapability &= ~AST_RTP_DTMF;
}
r->promiscredir = p->promiscredir;
r->usereqphone = p->usereqphone;
strncpy(r->context, p->context,sizeof(r->context)-1);
if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) &&
(!p->maxms || ((p->lastms >= 0) && (p->lastms <= p->maxms)))) {
@ -3623,6 +3628,34 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, char *cmd, c
char tmp[80];
char iabuf[INET_ADDRSTRLEN];
char *l = default_callerid, *n=NULL;
int x;
char urioptions[256]="";
if (p->usereqphone) {
char onlydigits = 1;
x=0;
/* Test p->username against allowed characters in AST_DIGIT_ANY
If it matches the allowed characters list, then sipuser = ";user=phone"
If not, then sipuser = ""
*/
/* + is allowed in first position in a tel: uri */
if (p->username && p->username[0] == '+')
x=1;
for (;x<strlen(p->username);x++) {
if (!strchr(AST_DIGIT_ANY, p->username[x])) {
onlydigits = 0;
break;
}
}
/* If we have only digits, add ;user=phone to the uri */
if (onlydigits)
strcpy(urioptions, ";user=phone");
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", cmd);
@ -3655,14 +3688,14 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, char *cmd, c
/* Otherwise, use the username while waiting for registration */
} else if (!ast_strlen_zero(p->username)) {
if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
snprintf(invite, sizeof(invite), "sip:%s@%s:%d",p->username, p->tohost, ntohs(p->sa.sin_port));
snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s",p->username, p->tohost, ntohs(p->sa.sin_port), urioptions);
} else {
snprintf(invite, sizeof(invite), "sip:%s@%s",p->username, p->tohost);
snprintf(invite, sizeof(invite), "sip:%s@%s%s",p->username, p->tohost, urioptions);
}
} else if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) {
snprintf(invite, sizeof(invite), "sip:%s:%d", p->tohost, ntohs(p->sa.sin_port));
snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions);
} else {
snprintf(invite, sizeof(invite), "sip:%s", p->tohost);
snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions);
}
strncpy(p->uri, invite, sizeof(p->uri) - 1);
/* If there is a VXML URL append it to the SIP URL */
@ -5742,6 +5775,7 @@ static int sip_show_peer(int fd, int argc, char *argv[])
ast_cli(fd, " ACL : %s\n", (peer->ha?"Yes":"No"));
ast_cli(fd, " CanReinvite : %s\n", (peer->canreinvite?"Yes":"No"));
ast_cli(fd, " PromiscRedir : %s\n", (peer->promiscredir?"Yes":"No"));
ast_cli(fd, " User=Phone : %s\n", (peer->usereqphone?"Yes":"No"));
/* - is enumerated */
ast_cli(fd, " DTMFmode : ");
@ -8271,6 +8305,7 @@ static struct sip_peer *temp_peer(char *name)
peer->canreinvite = global_canreinvite;
peer->dtmfmode = global_dtmfmode;
peer->promiscredir = global_promiscredir;
peer->usereqphone = global_usereqphone;
peer->nat = global_nat;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
@ -8338,6 +8373,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
peer->addr.sin_family = AF_INET;
peer->defaddr.sin_family = AF_INET;
peer->expiry = expiry;
peer->usereqphone = global_usereqphone;
}
peer->prefs = prefs;
oldha = peer->ha;
@ -8380,6 +8416,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
strncpy(peer->context, v->value, sizeof(peer->context)-1);
else if (!strcasecmp(v->name, "fromdomain"))
strncpy(peer->fromdomain, v->value, sizeof(peer->fromdomain)-1);
else if (!strcasecmp(v->name, "usereqphone"))
peer->usereqphone = ast_true(v->value);
else if (!strcasecmp(v->name, "promiscredir"))
peer->promiscredir = ast_true(v->value);
else if (!strcasecmp(v->name, "fromuser"))
@ -8603,6 +8641,8 @@ static int reload_config(void)
strncpy(default_useragent, v->value, sizeof(default_useragent)-1);
ast_log(LOG_DEBUG, "Setting User Agent Name to %s\n",
default_useragent);
} else if (!strcasecmp(v->name, "usereqphone")) {
global_usereqphone = ast_true(v->value);
} else if (!strcasecmp(v->name, "relaxdtmf")) {
relaxdtmf = ast_true(v->value);
} else if (!strcasecmp(v->name, "promiscredir")) {

3
configs/sip.conf.sample
Unescape View File

@ -74,6 +74,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;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 SIP is incapable
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
; of performing a "hairpin" call.
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
@ -189,6 +191,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;[grandstream1]
;type=friend ; either "friend" (peer+user), "peer" or "user"

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