Module: sip-router
Branch: master
Commit: 408983f8a1403e844d679300f0c1ee2ece3cbd6c
URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=408983f…
Author: Daniel-Constantin Mierla <miconda(a)gmail.com>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: Wed Aug 18 10:48:16 2010 +0200
kamailio.cfg: updated users list email address
- fixed small typo
---
etc/kamailio.cfg | 4 ++--
1 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/etc/kamailio.cfg b/etc/kamailio.cfg
index 98fa69d..e4de392 100644
--- a/etc/kamailio.cfg
+++ b/etc/kamailio.cfg
@@ -4,7 +4,7 @@
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
-# Direct your questions about this file to: <users(a)lists.kamailio.org>
+# Direct your questions about this file to: <sr-users(a)lists.sip-router.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
@@ -87,7 +87,7 @@ children=4
#disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
- based on revers DNS on IPs (default on) */
+ based on reverse DNS on IPs (default on) */
#auto_aliases=no
/* add local domain aliases */
Hello,
the path to modules can be specified via command line option -L or cfg
parameter loadpath (aliased as mpath for K backward compat). loadpath
overwrites the value given by -L.
The question is whether the value given via -L shouldn't be effective
all the time (i.e., loadpath cfg parameter ignored when -L is given)
since it is via command line parameter -- my expectation is command line
options should have higher priority. Changing it now will break backward
compatibility (if anyone used it like now) while in testing, but if all
think is a bug, then can be fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com/
Module: sip-router
Branch: master
Commit: 5004a60ad8a9cd47a2c6a7fc6eae82c1b371f92a
URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=5004a60…
Author: Ovidiu Sas <osas(a)voipembedded.com>
Committer: Ovidiu Sas <osas(a)voipembedded.com>
Date: Tue Aug 17 13:10:33 2010 -0400
modules_k:rtpproxy - remove trailing whitespaces from documentation
---
modules_k/rtpproxy/doc/nathelper_admin.xml | 60 ++++++++++++++--------------
1 files changed, 30 insertions(+), 30 deletions(-)
diff --git a/modules_k/rtpproxy/doc/nathelper_admin.xml b/modules_k/rtpproxy/doc/nathelper_admin.xml
index d50665a..671dff1 100644
--- a/modules_k/rtpproxy/doc/nathelper_admin.xml
+++ b/modules_k/rtpproxy/doc/nathelper_admin.xml
@@ -21,7 +21,7 @@
via an rtpproxy.
</para>
<para>
- Known devices that get along over &nat;s with rtpproxy are ATAs
+ Known devices that get along over &nat;s with rtpproxy are ATAs
(as clients) and Cisco Gateways (since 12.2(T)) as servers. See <ulink
url="http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_gui…">
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_gui…"></ulink>
@@ -35,10 +35,10 @@
balancing/distribution and control/selection purposes.
</para>
<para>
- The module allows the definition of several sets of rtpproxies -
+ The module allows the definition of several sets of rtpproxies -
load-balancing will be performed over a set and the user has the
ability to choose what set should be used. The set is selected via
- its id - the id being defined along with the set. Refer to the
+ its id - the id being defined along with the set. Refer to the
<quote>rtpproxy_sock</quote> module parameter definition for syntax
description.
</para>
@@ -47,13 +47,13 @@
the weight of each rtpproxy from the set.
</para>
<para>
- The selection of the set is done from script prior using
+ The selection of the set is done from script prior using
[un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
functions - see the set_rtp_proxy_set() function.
</para>
<para>
- For backward compatibility reasons, a set with no id take by default
- the id 0. Also if no set is explicitly set before
+ For backward compatibility reasons, a set with no id take by default
+ the id 0. Also if no set is explicitly set before
[un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
the 0 id set will be used.
</para>
@@ -81,7 +81,7 @@
<section>
<title>External Libraries or Applications</title>
<para>
- The following libraries or applications must be installed before
+ The following libraries or applications must be installed before
running &kamailio; with this module loaded:
<itemizedlist>
<listitem>
@@ -99,7 +99,7 @@
<section>
<title><varname>rtpproxy_sock</varname> (string)</title>
<para>
- Definition of socket(s) used to connect to (a set) RTPProxy. It may
+ Definition of socket(s) used to connect to (a set) RTPProxy. It may
specify a UNIX socket or an IPv4/IPv6 UDP socket.
</para>
<para>
@@ -129,7 +129,7 @@ modparam("rtpproxy", "rtpproxy_sock",
<title><varname>rtpproxy_disable_tout</varname> (integer)</title>
<para>
Once RTPProxy was found unreachable and marked as disable, rtpproxy
- will not attempt to establish communication to RTPProxy for
+ will not attempt to establish communication to RTPProxy for
rtpproxy_disable_tout seconds.
</para>
<para>
@@ -241,12 +241,12 @@ modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
<function moreinfo="none">set_rtp_proxy_set()</function>
</title>
<para>
- Sets the Id of the rtpproxy set to be used for the next
+ Sets the Id of the rtpproxy set to be used for the next
[un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
command.
</para>
<para>
- This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
+ This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
</para>
<example>
@@ -264,7 +264,7 @@ force_rtp_proxy();
<function moreinfo="none">force_rtp_proxy([flags [, ip_address]])</function>
</title>
<para>
- Rewrites &sdp; body to ensure that media is passed through
+ Rewrites &sdp; body to ensure that media is passed through
an &rtp; proxy. It can have optional parameters to force additional
features. If ip_address is provided, it will be used to replace the
one in SDP.
@@ -298,7 +298,7 @@ force_rtp_proxy();
</para></listitem>
<listitem><para>
<emphasis>l</emphasis> - force <quote>lookup</quote>, that is,
- only rewrite SDP when corresponding session is already exists
+ only rewrite SDP when corresponding session is already exists
in the RTP proxy. By default is on when the session is to be
completed (reply in non-swap or ACK in swap mode).
</para></listitem>
@@ -318,29 +318,29 @@ force_rtp_proxy();
the 'w' flag for clients behind NAT! See also above notes!
</para></listitem>
<listitem><para>
- <emphasis>f</emphasis> - instructs rtpproxy to ignore marks
- inserted by another rtpproxy in transit to indicate that the
- session is already goes through another proxy. Allows creating
+ <emphasis>f</emphasis> - instructs rtpproxy to ignore marks
+ inserted by another rtpproxy in transit to indicate that the
+ session is already goes through another proxy. Allows creating
chain of proxies.
</para></listitem>
<listitem><para>
- <emphasis>r</emphasis> - flags that IP address in SDP should
- be trusted. Without this flag, rtpproxy ignores address in
- the SDP and uses source address of the SIP message as media
+ <emphasis>r</emphasis> - flags that IP address in SDP should
+ be trusted. Without this flag, rtpproxy ignores address in
+ the SDP and uses source address of the SIP message as media
address which is passed to the RTP proxy.
</para></listitem>
<listitem><para>
- <emphasis>o</emphasis> - flags that IP from the origin
+ <emphasis>o</emphasis> - flags that IP from the origin
description (o=) should be also changed.
</para></listitem>
<listitem><para>
- <emphasis>c</emphasis> - flags to change the session-level
- SDP connection (c=) IP if media-description also includes
+ <emphasis>c</emphasis> - flags to change the session-level
+ SDP connection (c=) IP if media-description also includes
connection information.
</para></listitem>
<listitem><para>
- <emphasis>s</emphasis> - flags to swap creation with
- confirmation between requests and replies. By default, a
+ <emphasis>s</emphasis> - flags to swap creation with
+ confirmation between requests and replies. By default, a
request creates the RTP session and a reply confirms it. If
swapped, a reply will create the RTP session and a request
will confirm it. The flag is considered depreciated and
@@ -348,7 +348,7 @@ force_rtp_proxy();
rtpproxy_offer() or rtpproxy_answer() instead.
</para></listitem>
<listitem><para>
- <emphasis>w</emphasis> - flags that for the UA from which
+ <emphasis>w</emphasis> - flags that for the UA from which
message is received, support symmetric RTP must be forced.
</para></listitem>
<listitem><para>
@@ -371,7 +371,7 @@ force_rtp_proxy();
</para></listitem>
</itemizedlist>
<para>
- This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
+ This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
</para>
<example>
@@ -568,7 +568,7 @@ unforce_rtp_proxy();
<function moreinfo="none">start_recording()</function>
</title>
<para>
- This command will send a signal to the RTP-Proxy to record
+ This command will send a signal to the RTP-Proxy to record
the RTP stream on the RTP-Proxy.
</para>
<para>
@@ -600,7 +600,7 @@ start_recording();
Disables it if a zero value is given.
</para>
<para>
- The first parameter is the rtp proxy url (exactly as defined in
+ The first parameter is the rtp proxy url (exactly as defined in
the config file).
</para>
<para>
@@ -624,7 +624,7 @@ $ &ctltool; fifo nh_enable_rtpp udp:192.168.2.133:8081 0
<section>
<title><function moreinfo="none">nh_show_rtpp</function></title>
<para>
- Displays all the rtp proxies and their information: set and
+ Displays all the rtp proxies and their information: set and
status (disabled or not, weight and recheck_ticks).
</para>
<para>
@@ -635,7 +635,7 @@ $ &ctltool; fifo nh_enable_rtpp udp:192.168.2.133:8081 0
<function moreinfo="none">nh_show_rtpp</function> usage</title>
<programlisting format="linespecific">
...
-$ &ctltool; fifo nh_show_rtpp
+$ &ctltool; fifo nh_show_rtpp
...
</programlisting>
</example>
Hello,
I committed in master branch (to be 3.1.0) code that allows to set uri,
username, domain and display name for To and From headers using
assignments to their respective PVs in configuration file. For example:
$fu = "sip:anonymous@invalid";
$fn = '"Jon Doe"';
$tU = "+123455678";
Assignment of each such attribute should be done only once, otherwise
you get concatenated values since it uses the internal lump system.
Therefore, doing an update is not visible immediately in config unless
you do use msg_apply_changes().
Also, use it carefully, in case you have sipv1 devices in your network
then it can break dialog matching. Anyhow changing content of From and
To headers was possible with remove_hf()/append_hf() or subst()
functions, the new feature comes just to ease writing the config when
one needs it.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com/