[Pkg-voip-commits] r9410 - /asterisk/branches/squeeze/debian/patches/AST-2011-013
tzafrir at alioth.debian.org
tzafrir at alioth.debian.org
Sat Dec 17 12:05:27 UTC 2011
Author: tzafrir
Date: Sat Dec 17 12:05:27 2011
New Revision: 9410
URL: http://svn.debian.org/wsvn/pkg-voip/?sc=1&rev=9410
Log:
Refresh patch AST-2011-013
Modified:
asterisk/branches/squeeze/debian/patches/AST-2011-013
Modified: asterisk/branches/squeeze/debian/patches/AST-2011-013
URL: http://svn.debian.org/wsvn/pkg-voip/asterisk/branches/squeeze/debian/patches/AST-2011-013?rev=9410&op=diff
==============================================================================
--- asterisk/branches/squeeze/debian/patches/AST-2011-013 (original)
+++ asterisk/branches/squeeze/debian/patches/AST-2011-013 Sat Dec 17 12:05:27 2011
@@ -26,8 +26,6 @@
configs/sip.conf.sample | 17 +++++++++--------
3 files changed, 46 insertions(+), 20 deletions(-)
-diff --git a/CHANGES b/CHANGES
-index f200a60..63ed23b 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,18 @@
@@ -49,11 +47,9 @@
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
------------------------------------------------------------------------------
-diff --git a/channels/chan_sip.c b/channels/chan_sip.c
-index 328643e..a9a5085 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
-@@ -24164,15 +24164,14 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
+@@ -23568,15 +23568,14 @@ static int handle_common_options(struct
}
} else if (!strcasecmp(v->name, "nat")) {
ast_set_flag(&mask[0], SIP_NAT);
@@ -77,7 +73,7 @@
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
-@@ -25124,6 +25123,15 @@ static int peer_markall_func(void *device, void *arg, int flags)
+@@ -24491,6 +24490,15 @@ static int peer_markall_func(void *devic
return 0;
}
@@ -93,7 +89,7 @@
/*! \brief Re-read SIP.conf config file
\note This function reloads all config data, except for
active peers (with registrations). They will only
-@@ -25338,9 +25349,10 @@ static int reload_config(enum channelreloadreason reason)
+@@ -24705,9 +24713,10 @@ static int reload_config(enum channelrel
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
@@ -107,7 +103,7 @@
/* Debugging settings, always default to off */
dumphistory = FALSE;
-@@ -25993,6 +26005,7 @@ static int reload_config(enum channelreloadreason reason)
+@@ -25301,6 +25310,7 @@ static int reload_config(enum channelrel
}
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
if (peer) {
@@ -115,11 +111,9 @@
ao2_t_link(peers, peer, "link peer into peers table");
if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
-diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
-index 1eafdb6..e9abacc 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
-@@ -660,10 +660,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
+@@ -656,10 +656,18 @@ srvlookup=yes ; Enable
; The following settings are allowed (both globally and in individual sections):
;
; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
@@ -139,7 +133,7 @@
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
-@@ -990,12 +998,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
+@@ -982,12 +990,10 @@ srvlookup=yes ; Enable
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
@@ -152,7 +146,7 @@
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
-@@ -1030,7 +1036,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
+@@ -1022,7 +1028,6 @@ srvlookup=yes ; Enable
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
@@ -160,7 +154,7 @@
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
-@@ -1060,7 +1065,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
+@@ -1052,7 +1057,6 @@ srvlookup=yes ; Enable
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
@@ -168,7 +162,7 @@
;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
-@@ -1131,9 +1135,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
+@@ -1123,9 +1127,6 @@ srvlookup=yes ; Enable
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
@@ -178,6 +172,3 @@
;host=dynamic ; This device registers with us
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
---
-1.7.7.3
-
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