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authorMaxim Sobolev <sobomax@FreeBSD.org>2008-12-16 08:16:35 +0000
committerMaxim Sobolev <sobomax@FreeBSD.org>2008-12-16 08:16:35 +0000
commit7fa0f7ea18bb65b7106661be60fd563ba6fa3b4f (patch)
tree425e0d220acdcc79f771ffa6b939a74ec19750bb
parent8bdef2016466c4547b96c5528f735d01955f9467 (diff)
downloadports-7fa0f7ea18bb65b7106661be60fd563ba6fa3b4f.tar.gz
ports-7fa0f7ea18bb65b7106661be60fd563ba6fa3b4f.zip
Notes
Diffstat
-rw-r--r--net/asterisk/files/rtp_force_dtmf-nocodecnego.diff24
-rw-r--r--net/asterisk10/files/rtp_force_dtmf-nocodecnego.diff24
-rw-r--r--net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff24
-rw-r--r--net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff24
4 files changed, 48 insertions, 48 deletions
diff --git a/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff
index 776422b34645..9b1d521fa0e3 100644
--- a/net/asterisk/files/rtp_force_dtmf-nocodecnego.diff
+++ b/net/asterisk/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.
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.
diff --git a/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff
index 776422b34645..9b1d521fa0e3 100644
--- a/net/asterisk14/files/rtp_force_dtmf-nocodecnego.diff
+++ b/net/asterisk14/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.
diff --git a/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff
index 776422b34645..9b1d521fa0e3 100644
--- a/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff
+++ b/net/asterisk16/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.
generated by cgit v1.2.3 (git 2.25.1) at 2024-10-23 03:32:46 +0000