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authorjeremy <jeremy@f38db490-d61c-443f-a65b-d21fe96a405b>2004-12-21 00:07:56 +0000
committerjeremy <jeremy@f38db490-d61c-443f-a65b-d21fe96a405b>2004-12-21 00:07:56 +0000
commit9e8b32b39c3ee7a83f4d00febfe7d1d6e421516e (patch)
tree369252e4ce57f37a8e9e1d3ff7b800a667fb016a /channels/h323/chan_h323.h
parent625e2c693e6230c2e0aafd11089fafbd09229d19 (diff)
Fix G.729 name annoucement to Open H.323 and inform Asterisk's RTP stack specifically which codec we are expecting.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@4508 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/h323/chan_h323.h')
-rwxr-xr-xchannels/h323/chan_h323.h4
1 files changed, 3 insertions, 1 deletions
diff --git a/channels/h323/chan_h323.h b/channels/h323/chan_h323.h
index fca116ea9..a59da67fc 100755
--- a/channels/h323/chan_h323.h
+++ b/channels/h323/chan_h323.h
@@ -99,6 +99,8 @@ typedef struct call_details {
char *call_source_name;
char *call_source_e164;
char *call_dest_e164;
+ int presentation;
+ int screening;
char *sourceIp;
} call_details_t;
@@ -119,7 +121,7 @@ extern on_rtp_cb on_external_rtp_create;
/* This is a callback prototype function, called to send
the remote IP and RTP port from H.323 to Asterisk */
-typedef void (*start_rtp_cb)(unsigned int, const char *, int, const char *);
+typedef void (*start_rtp_cb)(unsigned int, const char *, int, const char *, int);
extern start_rtp_cb on_start_rtp_channel;
/* This is a callback that happens when call progress is