[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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