[Asterisk-code-review] chan sip: Change sip get codec() to return correct codec list (asterisk[13])
Vitezslav Novy
asteriskteam at digium.com
Fri May 12 04:30:17 CDT 2017
Vitezslav Novy has uploaded a new change for review. ( https://gerrit.asterisk.org/5620 )
Change subject: chan_sip: Change sip_get_codec() to return correct codec list
......................................................................
chan_sip: Change sip_get_codec() to return correct codec list
Return cahnnel nativeformats to fix bridge technology selection process.
Same approach as in pjsip module.
ASTERISK-26143
Reported-by: Henning Holtschneider
Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 3 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/20/5620/1
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2c5d3c3..43084ce 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -33481,9 +33481,7 @@
static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
- struct sip_pvt *p = ast_channel_tech_pvt(chan);
-
- ast_format_cap_append_from_cap(result, !ast_format_cap_count(p->peercaps) ? p->caps : p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
+ ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
}
static struct ast_rtp_glue sip_rtp_glue = {
--
To view, visit https://gerrit.asterisk.org/5620
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Vitezslav Novy <a1 at vnovy.net>
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