From 09844a7f1a8745254f6f8fb5ac733f4934769c37 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 1 Feb 2007 21:54:28 +0000 Subject: [PATCH] Merged revisions 53095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 127b2c3260..5ab633cb29 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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,