Add documentation to the ability to retrieve the source port of a SIP call.

(belongs with r419970)

ASTERISK-24040 #close
Patches:
	func_channel.c.diff uploaded by dtryba

Review: https://reviewboard.asterisk.org/r/3781/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/funcs/func_channel.c b/funcs/func_channel.c
index 3f756f0b5b1cf48d48e07109833ef901f3a8dab3..814960ac233ed32495df9f2b4cf9fe72c8b9ce21 100644 (file)
--- a/funcs/func_channel.c
+++ b/funcs/func_channel.c
@@ -213,6 +213,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                                        <enum name="recvip">
                                                <para>R/O Get the source IP address of the peer.</para>
                                        </enum>
+                                       <enum name="recvport">
+                                               <para>R/O Get the source port of the peer.</para>
+                                       </enum>
                                        <enum name="from">
                                                <para>R/O Get the URI from the From: header.</para>
                                        </enum>