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Merged revisions 53095 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines

Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) 

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2007-02-01 21:54:28 +00:00
parent 51bcb9d3be
commit 09844a7f1a
1 changed files with 7 additions and 7 deletions
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14
channels/chan_sip.c
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@@ -558,7 +558,6 @@ static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction schem
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static int noncodeccapability = AST_RTP_DTMF;
/* Object counters */
static int suserobjs = 0; /*!< Static users */
@@ -943,6 +942,7 @@ static struct sip_pvt {
int peercapability; /*!< Supported peer capability */
int prefcodec; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
struct t38properties t38; /*!< T38 settings */
@@ -5104,7 +5104,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);
newnoncodeccapability = noncodeccapability & peernoncodeccapability;
newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
@@ -5118,7 +5118,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
}
@@ -5137,9 +5137,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->noncodeccapability = newnoncodeccapability; /* DTMF capabilities */
p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
@@ -6276,7 +6276,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
if (!(p->noncodeccapability & x))
if (!(p->jointnoncodeccapability & x))
continue;
add_noncodec_to_sdp(p, x, 8000,