kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-18 18:58:22 +00:00
Code Issues Packages Projects Releases Wiki Activity

Fix SIP ACK for BYE (bug #3087)

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-0@4500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2004-12-20 01:48:20 +00:00
parent 00f2912aaa
commit 6f60468d3d
1 changed files with 13 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
channels/chan_sip.c
View File

@@ -3100,6 +3100,7 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, stru
return 0;
}
/*--- reqprep: Initialize a SIP request packet ---*/
static int reqprep(struct sip_request *req, struct sip_pvt *p, char *msg, int seqno, int newbranch)
{
struct sip_request *orig = &p->initreq;
@@ -3135,9 +3136,12 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, char *msg, int se
c = p->okcontacturi;
else
c = p->initreq.rlPart2;
} else if (!ast_strlen_zero(p->okcontacturi)) {
c = p->okcontacturi; /* Use for BYE or REINVITE */
} else if (!ast_strlen_zero(p->uri)) {
c = p->uri;
} else {
/* We have no URI, use To: or From: header as URI (depending on direction) */
if (p->outgoing)
strncpy(stripped, get_header(orig, "To"), sizeof(stripped) - 1);
else
@@ -3569,7 +3573,7 @@ static int determine_firstline_parts( struct sip_request *req ) {
e++;
if( !*e ) { return -1; }
}
req->rlPart2= e;
req->rlPart2= e; /* URI */
if( ( e= strrchr( req->rlPart2, 'S' ) ) == NULL ) {
return -1;
}
@@ -3583,9 +3587,12 @@ static int determine_firstline_parts( struct sip_request *req ) {
return 1;
}
/* transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/
/* A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
INVITE that opened the SIP dialogue */
/*--- transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/
/* A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
INVITE that opened the SIP dialogue
We reinvite so that the audio stream (RTP) go directly between
the SIP UAs. SIP Signalling stays with * in the path.
*/
static int transmit_reinvite_with_sdp(struct sip_pvt *p)
{
struct sip_request req;
@@ -6563,6 +6570,7 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
}
}
/*--- check_pendings: Check pending actions on SIP call ---*/
static void check_pendings(struct sip_pvt *p)
{
/* Go ahead and send bye at this point */
@@ -8865,6 +8873,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
return rtp;
}
/*--- sip_set_rtp_peer: Set the RTP peer for this call ---*/
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
{
struct sip_pvt *p;