Two other trunk build fixes (reported by seanbright on #asterisk-dev)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 1948bc5..b0aa8c0 100644 (file)
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -1517,7 +1517,7 @@ static unsigned char compress_subclass(format_t subclass)
        for (x = 0; x < IAX_MAX_SHIFT; x++) {
                if (subclass & (1LL << x)) {
                        if (power > -1) {
-                               ast_log(LOG_WARNING, "Can't compress subclass %Ld\n", subclass);
+                               ast_log(LOG_WARNING, "Can't compress subclass %Ld\n", (long long) subclass);
                                return 0;
                        } else
                                power = x;

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 438df8f..b2b65dc 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10194,7 +10194,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
 
 
        if (debug)
-               ast_verbose("Adding codec 0x%Lx (%s) to SDP\n", codec, ast_getformatname(codec));
+               ast_verbose("Adding codec 0x%Lx (%s) to SDP\n", (long long) codec, ast_getformatname(codec));
        if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
                return;