[Asterisk-cvs] asterisk/channels chan_sip.c,1.378,1.379
markster at lists.digium.com
markster at lists.digium.com
Sun May 9 16:56:00 CDT 2004
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv11412/channels
Modified Files:
chan_sip.c
Log Message:
Add SIPUSERAGENT, SIPCALLID, and fix some formatting (bug #1593)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.378
retrieving revision 1.379
diff -u -d -r1.378 -r1.379
--- chan_sip.c 9 May 2004 20:46:18 -0000 1.378
+++ chan_sip.c 9 May 2004 21:05:43 -0000 1.379
@@ -218,81 +218,82 @@
};
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
char *cseq;
char *from;
char *e;
+ char *useragent;
struct ast_channel *c=NULL;
struct ast_channel *transfer_to;
int seqno;
@@ -5820,12 +5854,17 @@
/* 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 */
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