[Asterisk-code-review] chan sip: Change sip pvt.get codec() to return correct codec... (asterisk[13])

Vitezslav Novy asteriskteam at digium.com
Fri May 12 03:16:08 CDT 2017


Vitezslav Novy has uploaded a new change for review. ( https://gerrit.asterisk.org/5618 )

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/18/5618/1

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/5618
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I226952bc4d5f65f33eb9ba44c08516529afe05f0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Vitezslav Novy <a1 at vnovy.net>



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