diff options
| author | sobomax <sobomax@FreeBSD.org> | 2008-12-16 16:16:35 +0800 |
|---|---|---|
| committer | sobomax <sobomax@FreeBSD.org> | 2008-12-16 16:16:35 +0800 |
| commit | 0d12358d0be3fbd06eb89f6414c5aff5d3e80afd (patch) | |
| tree | 770f8a242ad2969fb5db6d3f5c1b608c7e42dbad /net/asterisk10 | |
| parent | e805f2fdb652aab0bb34e9a15054618c16bc34aa (diff) | |
| download | freebsd-ports-gnome-0d12358d0be3fbd06eb89f6414c5aff5d3e80afd.tar.gz freebsd-ports-gnome-0d12358d0be3fbd06eb89f6414c5aff5d3e80afd.tar.zst freebsd-ports-gnome-0d12358d0be3fbd06eb89f6414c5aff5d3e80afd.zip | |
Fix build with misc patches on and codec negotiation patch off. This
combination is not enabled by default so that no PORTREVISION bump.
Reported by: Peter Beckman
Diffstat (limited to 'net/asterisk10')
| -rw-r--r-- | net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff index 776422b34645..9b1d521fa0e3 100644 --- a/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff +++ b/net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff @@ -1,6 +1,6 @@ ---- channels/chan_sip.c.orig 2008-01-31 21:52:49.000000000 +0200 -+++ channels/chan_sip.c 2008-03-14 17:50:57.000000000 +0200 -@@ -556,6 +556,9 @@ +--- channels/chan_sip.c.orig 2008-09-09 00:10:10.000000000 +0300 ++++ channels/chan_sip.c 2008-12-12 17:02:05.000000000 +0200 +@@ -557,6 +557,9 @@ static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */ static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */ static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */ @@ -10,7 +10,7 @@ static int compactheaders; /*!< send compact sip headers */ static int recordhistory; /*!< Record SIP history. Off by default */ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */ -@@ -5392,6 +5395,13 @@ +@@ -5504,6 +5507,13 @@ /* Now gather all of the codecs that we are asked for: */ ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability); @@ -24,7 +24,7 @@ ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability); newjointcapability = p->capability & (peercapability | vpeercapability); -@@ -16833,6 +16843,9 @@ +@@ -17440,6 +17450,9 @@ global_matchexterniplocally = FALSE; @@ -34,7 +34,7 @@ /* Copy the default jb config over global_jbconf */ memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf)); -@@ -16889,6 +16902,18 @@ +@@ -17496,6 +17509,18 @@ } } else if (!strcasecmp(v->name, "vmexten")) { ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten)); @@ -53,11 +53,11 @@ } else if (!strcasecmp(v->name, "rtptimeout")) { if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) { ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); ---- configs/sip.conf.sample.orig 2008-03-12 17:57:19.000000000 +0200 -+++ configs/sip.conf.sample 2008-03-12 18:13:03.000000000 +0200 -@@ -53,6 +53,12 @@ - ; and multiline formatted headers for strict - ; SIP compatibility (defaults to "no") +--- configs/sip.conf.sample.orig 2008-08-16 01:33:42.000000000 +0300 ++++ configs/sip.conf.sample 2008-12-12 17:03:11.000000000 +0200 +@@ -49,6 +49,12 @@ + ; and multiline formatted headers for strict + ; SIP compatibility (defaults to "no") +;rtp_force_dtmf_relay=no ; Enable RFC2833 DTMFs to be sent even if peer + ; hasn't announced support for it. Default: no @@ -65,6 +65,6 @@ +;rtp_force_dtmf_relay_pt=101 ; RTP payload type value for enforced RFC2833 + ; DTMFs. Default: 101 + - ; See doc/README.tos for a description of these parameters. + ; See doc/ip-tos.txt for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. |