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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2004-05-09 21:05:43 +00:00
parent 62eef7c695
commit 6ead43beb1
2 changed files with 120 additions and 79 deletions
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197
channels/chan_sip.c
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@@ -218,81 +218,82 @@ struct sip_route {
};
static struct sip_pvt {
ast_mutex_t lock; /* Channel private lock */
char callid[80]; /* Global CallID */
char randdata[80]; /* Random data */
unsigned int ocseq; /* Current outgoing seqno */
unsigned int icseq; /* Current incoming seqno */
unsigned int callgroup;
unsigned int pickupgroup;
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 jointcapability; /* Supported capability at both ends */
int prefcodec; /* Preferred codec (outbound only) */
ast_mutex_t lock; /* Channel private lock */
char callid[80]; /* Global CallID */
char randdata[80]; /* Random data */
unsigned int ocseq; /* Current outgoing seqno */
unsigned int icseq; /* Current incoming seqno */
unsigned int callgroup; /* Call group */
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 (codec) */
int jointcapability; /* Supported capability at both ends (codecs ) */
int prefcodec; /* Preferred codec (outbound only) */
int noncodeccapability;
int outgoing; /* Outgoing or incoming call? */
int authtries; /* Times we've tried to authenticate */
int insecure; /* Don't check source port/ip */
int expiry; /* How long we take to expire */
int branch; /* One random number */
int canreinvite; /* Do we support reinvite */
int ringing; /* Have sent 180 ringing */
int progress; /* Have sent 183 message progress */
int tag; /* Another random number */
int nat; /* Whether to try to support NAT */
int sessionid; /* SDP Session ID */
int sessionversion; /* SDP Session Version */
struct sockaddr_in sa; /* Our peer */
struct sockaddr_in redirip; /* Where our RTP should be going if not to us */
int outgoing; /* Outgoing or incoming call? */
int authtries; /* Times we've tried to authenticate */
int insecure; /* Don't check source port/ip */
int expiry; /* How long we take to expire */
int branch; /* One random number */
int canreinvite; /* Do we support reinvite */
int ringing; /* Have sent 180 ringing */
int progress; /* Have sent 183 message progress */
int tag; /* Another random number */
int nat; /* Whether to try to support NAT */
int sessionid; /* SDP Session ID */
int sessionversion; /* SDP Session Version */
struct sockaddr_in sa; /* Our peer */
struct sockaddr_in redirip; /* Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /* Where our Video RTP should be going if not to us */
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 */
struct sockaddr_in recv; /* Received as */
struct in_addr ourip; /* Our IP */
struct ast_channel *owner; /* Who owns us */
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 */
struct sip_pvt *refer_call; /* Call we are referring */
struct sip_route *route; /* Head of linked list of routing steps (fm Record-Route) */
int route_persistant; /* Is this the "real" route? */
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 */
struct sip_pvt *refer_call; /* Call we are referring */
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 context[AST_MAX_EXTENSION];
char from[256]; /* The From: header */
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 */
char rdnis[256]; /* Referring DNIS */
char theirtag[256]; /* Their tag */
char username[256];
char peername[256];
char authname[256]; /* Who we use for authentication */
char uri[256]; /* Original requested URI */
char authname[256]; /* Who we use for authentication */
char uri[256]; /* Original requested URI */
char peersecret[256];
char peermd5secret[256];
char callerid[256]; /* Caller*ID */
char callerid[256]; /* Caller*ID */
int restrictcid; /* hide presentation from remote user */
char via[256];
char accountcode[20]; /* Account code */
char our_contact[256]; /* Our contact header */
char realm[256]; /* Authorization realm */
char nonce[256]; /* Authorization nonce */
char opaque[256]; /* Opaque nonsense */
char accountcode[20]; /* Account code */
char our_contact[256]; /* Our contact header */
char realm[256]; /* Authorization realm */
char nonce[256]; /* Authorization nonce */
char opaque[256]; /* Opaque nonsense */
char qop[80]; /* Quality of Protection, since SIP wasn't complicated enough yet. */
char domain[256]; /* Authorization nonce */
char lastmsg[256]; /* Last Message sent/received */
int amaflags; /* AMA Flags */
int pendinginvite; /* Any pending invite */
int pendingbye; /* Need to send bye after we ack? */
int gotrefer; /* Got a refer? */
struct sip_request initreq; /* Initial request */
char domain[256]; /* Authorization nonce */
char lastmsg[256]; /* Last Message sent/received */
int amaflags; /* AMA Flags */
int pendinginvite; /* Any pending invite */
int pendingbye; /* Need to send bye after we ack? */
int gotrefer; /* Got a refer? */
struct sip_request initreq; /* Initial request */
int maxtime; /* Max time for first response */
int initid; /* Auto-congest ID if appropriate */
int autokillid; /* Auto-kill ID */
int maxtime; /* Max time for first response */
int initid; /* Auto-congest ID if appropriate */
int autokillid; /* Auto-kill ID */
int subscribed;
int stateid;
@@ -301,12 +302,12 @@ static struct sip_pvt {
int dtmfmode;
struct ast_dsp *vad;
struct sip_peer *peerpoke; /* If this calls is to poke a peer, which one */
struct sip_registry *registry; /* If this is a REGISTER call, to which registry */
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_peer *peerpoke; /* If this calls is to poke a peer, which one */
struct sip_registry *registry; /* If this is a REGISTER call, to which registry */
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; /* 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;
@@ -4977,18 +4994,34 @@ 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, "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, "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, "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");
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");
strcpy(tmp, "");
if (cur->dtmfmode & SIP_DTMF_RFC2833)
strcat(tmp, "rfc2833 ");
@@ -4996,7 +5029,7 @@ static int sip_show_channel(int fd, int argc, char *argv[])
strcat(tmp, "info ");
if (cur->dtmfmode & SIP_DTMF_INBAND)
strcat(tmp, "inband ");
ast_cli(fd, "DTMF Mode: %s\n\n", tmp);
ast_cli(fd, " DTMF Mode: %s\n\n", tmp);
}
cur = cur->next;
}
@@ -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;
@@ -5820,12 +5854,17 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
/* Get the command */
cseq += len;
/* Determine the request URI for sip, sips or tel URIs */
/* Determine the request URI for sip, sips or tel URIs */
if( determine_firstline_parts( req ) < 0 ) {
return -1;
}
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 */

2
doc/README.variables
View File

@@ -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