Add SIPUSERAGENT, SIPCALLID, and fix some formatting (bug #1593)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2937 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -223,13 +223,13 @@ static struct sip_pvt {
 	char randdata[80];			/* Random data */
 	unsigned int ocseq;			/* Current outgoing seqno */
 	unsigned int icseq;			/* Current incoming seqno */
-	unsigned int callgroup;
+	unsigned int callgroup;			/* Call group */
-	unsigned int pickupgroup;
+	unsigned int pickupgroup;		/* Pickup group */
 	int lastinvite;				/* Last Cseq of invite */
 	int alreadygone;			/* Whether or not we've already been destroyed by or peer */
 	int needdestroy;			/* if we need to be destroyed */
-	int capability;						/* Special capability */
+	int capability;				/* Special capability (codec) */
-	int jointcapability;				/* Supported capability at both ends */
+	int jointcapability;			/* Supported capability at both ends (codecs ) */
 	int prefcodec;				/* Preferred codec (outbound only) */
 	int noncodeccapability;
 	int outgoing;				/* Outgoing or incoming call? */
@@ -250,7 +250,7 @@ static struct sip_pvt {
 	struct sockaddr_in recv;		/* Received as */
 	struct in_addr ourip;			/* Our IP */
 	struct ast_channel *owner;		/* Who owns us */
-	char exten[AST_MAX_EXTENSION];		/* Extention where to start */
+	char exten[AST_MAX_EXTENSION];		/* Extension where to start */
 	char refer_to[AST_MAX_EXTENSION];	/* Place to store REFER-TO extension */
 	char referred_by[AST_MAX_EXTENSION];	/* Place to store REFERRED-BY extension */
 	char refer_contact[AST_MAX_EXTENSION];	/* Place to store Contact info from a REFER extension */
@@ -258,12 +258,13 @@ static struct sip_pvt {
 	struct sip_route *route;		/* Head of linked list of routing steps (fm Record-Route) */
 	int route_persistant;			/* Is this the "real" route? */
 	char remote_party_id[256];
-	char from[256];
+	char from[256];				/* The From: header */
-	char context[AST_MAX_EXTENSION];
+	char useragent[256];			/* User agent in SIP request */
+	char context[AST_MAX_EXTENSION];	/* Context for this call */
 	char fromdomain[AST_MAX_EXTENSION];	/* Domain to show in the from field */
 	char fromuser[AST_MAX_EXTENSION];	/* Domain to show in the user field */
 	char tohost[AST_MAX_EXTENSION];		/* Host we should put in the "to" field */
-	char language[MAX_LANGUAGE];
+	char language[MAX_LANGUAGE];		/* Default language for this call */
 	char musicclass[MAX_LANGUAGE];          /* Music on Hold class */
 	char rdnis[256];			/* Referring DNIS */
 	char theirtag[256];			/* Their tag */
@@ -306,7 +307,7 @@ static struct sip_pvt {
 	struct ast_rtp *rtp;			/* RTP Session */
 	struct ast_rtp *vrtp;			/* Video RTP session */
 	struct sip_pkt *packets;		/* Packets scheduled for re-transmission */
-	struct sip_pvt *next;
+	struct sip_pvt *next;			/* Next call in chain */
 } *iflist = NULL;
 
 #define FLAG_RESPONSE (1 << 0)
@@ -334,6 +335,7 @@ struct sip_user {
 	char accountcode[20];
 	char language[MAX_LANGUAGE];
 	char musicclass[MAX_LANGUAGE];  /* Music on Hold class */
+	char useragent[256];		/* User agent in SIP request */
 	unsigned int callgroup;
 	unsigned int pickupgroup;
 	int nat;
@@ -365,6 +367,7 @@ struct sip_peer {
 	char mailbox[AST_MAX_EXTENSION];
 	char language[MAX_LANGUAGE];
 	char musicclass[MAX_LANGUAGE];  /* Music on Hold class */
+	char useragent[256];		/* User agent in SIP request */
 	int lastmsgssent;
 	time_t	lastmsgcheck;
 	int dynamic;
@@ -1661,6 +1664,12 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 		if (!ast_strlen_zero(i->domain)) {
 			pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
 		}
+		if (!ast_strlen_zero(i->useragent)) {
+			pbx_builtin_setvar_helper(tmp, "SIPUSERAGENT", i->useragent);
+		}
+		if (!ast_strlen_zero(i->callid)) {
+			pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
+		}
 		ast_setstate(tmp, state);
 		if (state != AST_STATE_DOWN) {
 			if (ast_pbx_start(tmp)) {
@@ -3733,6 +3742,12 @@ static int parse_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_req
 		if (option_verbose > 2)
 			ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d expires %d\n", p->name, inet_ntoa(p->addr.sin_addr), ntohs(p->addr.sin_port), expiry);
 	}
+
+	/* Save User agent */
+	strncpy(p->useragent, get_header(req, "User-Agent"),sizeof(p->useragent));
+	if (option_verbose > 2) 
+		ast_verbose(VERBOSE_PREFIX_3 "Saved useragent \"%s\" for peer %s\n",p->useragent,p->name);  
+
 	return 0;
 }
 
@@ -4856,6 +4871,7 @@ static int sip_show_peer(int fd, int argc, char *argv[])
 		else
 			strcpy(status, "UNKNOWN");
 		ast_cli(fd, "%s\n",status);
+ 		ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
 		ast_cli(fd,"\n");
 	} else {
 		ast_cli(fd,"Peer %s not found.\n", argv[3]);
@@ -4965,6 +4981,7 @@ static char *complete_sipch(char *line, char *word, int pos, int state)
 	return c;
 }
 
+/*--- sip_show_channel: Show details of one call ---*/
 static int sip_show_channel(int fd, int argc, char *argv[])
 {
 	struct sip_pvt *cur;
@@ -4977,15 +4994,31 @@ static int sip_show_channel(int fd, int argc, char *argv[])
 	cur = iflist;
 	while(cur) {
 		if (!strncasecmp(cur->callid, argv[3],len)) {
+			ast_cli(fd,"\n");
+			if (cur->subscribed)
+			   ast_cli(fd, "  * Subscription\n");
+			else
+			   ast_cli(fd, "  * SIP Call\n");
+			ast_cli(fd, "  Direction:              %s\n", cur->outgoing?"Outgoing":"Incoming");
 			ast_cli(fd, "  Call-ID:                %s\n", cur->callid);
 			ast_cli(fd, "  Our Codec Capability:   %d\n", cur->capability);
 			ast_cli(fd, "  Non-Codec Capability:   %d\n", cur->noncodeccapability);
 			ast_cli(fd, "  Joint Codec Capability:   %d\n", cur->jointcapability);
+			ast_cli(fd, "  Format                  %s\n", ast_getformatname(cur->owner ? cur->owner->nativeformats : 0) );
 			ast_cli(fd, "  Theoretical Address:    %s:%d\n", inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port));
 			ast_cli(fd, "  Received Address:       %s:%d\n", inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port));
 			ast_cli(fd, "  NAT Support:            %s\n", cur->nat ? "Yes" : "No");
 			ast_cli(fd, "  Our Tag:                %08d\n", cur->tag);
 			ast_cli(fd, "  Their Tag:              %s\n", cur->theirtag);
+			ast_cli(fd, "  SIP User agent:         %s\n", cur->useragent);
+			if (!ast_strlen_zero(cur->username))
+			   ast_cli(fd, "  Username:               %s\n", cur->username);
+			if (!ast_strlen_zero(cur->peername))
+			   ast_cli(fd, "  Peername:               %s\n", cur->peername);
+			if (!ast_strlen_zero(cur->uri))
+			   ast_cli(fd, "  Original uri:           %s\n", cur->uri);
+			if (!ast_strlen_zero(cur->callerid))
+			   ast_cli(fd, "  Caller-ID:              %s\n", cur->callerid);
 			ast_cli(fd, "  Need Destroy:           %d\n", cur->needdestroy);
 			ast_cli(fd, "  Last Message:           %s\n", cur->lastmsg);
 			ast_cli(fd, "  Route:                  %s\n", cur->route ? cur->route->hop : "N/A");
@@ -5796,6 +5829,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 	char *cseq;
 	char *from;
 	char *e;
+	char *useragent;
 	struct ast_channel *c=NULL;
 	struct ast_channel *transfer_to;
 	int seqno;
@@ -5827,6 +5861,11 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 	cmd= req->rlPart1;
 	e= req->rlPart2;
 
+	/* Save useragent of the client */
+	useragent = get_header(req, "User-Agent");
+	strncpy(p->useragent, useragent, sizeof(p->useragent)-1);
+
+	
 	if (strcasecmp(cmd, "SIP/2.0")) {
 		/* Request coming in */			
 		if (p->icseq && (p->icseq < seqno)) {

doc/README.variables

@@ -67,6 +67,8 @@ ${HANGUPCAUSE}	Asterisk hangup cause
 ${ACCOUNTCODE}  Account code (if specified)
 ${LANGUAGE}	Current language
 ${SIPDOMAIN}    SIP destination domain of an inbound call (if appropriate)
+${SIPUSERAGENT} SIP user agent 
+${SIPCALLID} 	SIP Call-ID: header verbatim (for logging or CDR matching)
 
 There are two reference modes - reference by value and reference by name. 
 To refer to a variable with its name (as an argument to a function that