diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 21:54:28 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 21:54:28 +0000 |
| commit | 0355d24c7bdd0431465a851f0f7e93c7d6673222 (patch) | |
| tree | f8862b17fc64de562277e387ef985df5c00ee8dd | |
| parent | 112d764088aebb364dd6cb45ceb8f1b7fbb95618 (diff) | |
Merged revisions 53095 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.4@53097 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 127b2c326..5ab633cb2 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, |