From 2bbda7a7c8706d60c94b2d47b33fa2b94e5cd7fc Mon Sep 17 00:00:00 2001 From: Tilghman Lesher Date: Wed, 4 Nov 2009 16:17:18 +0000 Subject: [PATCH] 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 --- channels/chan_iax2.c | 2 +- channels/chan_sip.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c index 1948bc51f2..b0aa8c024c 100644 --- 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 438df8f336..b2b65dc121 100644 --- 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;