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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4372 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 0540c49a94f288e71b41e296091a7375f143364a..7ebdc1b51b48f103dfaa961b1e2acf594aabfcb4 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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")) {

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 49194ab54cb19ef415e6b28369f12e1cafcb3f1d..eb201d4c1a17439f6b374bb014e05e6eace6ce98 100755 (executable)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -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"