[Asterisk-code-review] chan sip: Change sip pvt.get codec() to return correct codec... (asterisk[14])
Vitezslav Novy
asteriskteam at digium.com
Fri May 12 03:34:25 CDT 2017
Vitezslav Novy has uploaded a new change for review. ( https://gerrit.asterisk.org/5619 )
Change subject: chan_sip: Change sip_pvt.get_codec() to return correct codec list
......................................................................
chan_sip: Change sip_pvt.get_codec() to return correct codec list
Return nativeformatlist to fix bridge technology selection process.
Same approach as in pjsip module.
ASTERISK-26143 #close
Reported-by: Henning Holtschneider
Change-Id: I226952bc4d5f65f33eb9ba44c08516529afe05f0
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 3 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/19/5619/1
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2e386dd..ff2e5ba 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -33589,9 +33589,7 @@
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/5619
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I226952bc4d5f65f33eb9ba44c08516529afe05f0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Vitezslav Novy <a1 at vnovy.net>
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