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authorzuul <zuul@gerrit.asterisk.org>2016-03-31 17:35:23 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2016-03-31 17:35:23 -0500
commita04a62586fadb360d8f86248293f0c0191379c7c (patch)
treecf5edeb80bc87100667dcd305759788c2cbd71ca
parente2a8033d0966f9ce858bbe5a6dc9155ea48e095f (diff)
parent3f720155b7c8b3089c59cbfc78150aff4efc8240 (diff)
Merge "chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers."
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 09ab1a196..ffc2084a1 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13531,7 +13531,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
}
/* Finally our remaining audio/video codecs */
- for (x = 0; ast_test_flag(&p->flags[0], SIP_OUTGOING) && x < ast_format_cap_count(p->caps); x++) {
+ for (x = 0; p->outgoing_call && x < ast_format_cap_count(p->caps); x++) {
tmp_fmt = ast_format_cap_get_format(p->caps, x);
if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {