[asterisk-bugs] [JIRA] (ASTERISK-27448) Add ability to send progress inband by setting a channel variable instead of fixed config per endpoint
nappsoft (JIRA)
noreply at issues.asterisk.org
Wed Nov 29 05:32:07 CST 2017
nappsoft created ASTERISK-27448:
-----------------------------------
Summary: Add ability to send progress inband by setting a channel variable instead of fixed config per endpoint
Key: ASTERISK-27448
URL: https://issues.asterisk.org/jira/browse/ASTERISK-27448
Project: Asterisk
Issue Type: Improvement
Security Level: None
Components: Channels/chan_pjsip
Affects Versions: 13.18.2
Reporter: nappsoft
Severity: Minor
Attachments: progressinband.diff
I faced a problem with several voip switches when using 180 Ringing messages after announcements sent over 183 Session Progress with SDP. The solution would be to use inband_progress=yes but this way progress will be sent as rtp for each call, so as well in cases in which it is not necessary.
So my idea was to set a channel variable PROGRESSINBAND to 1 in situations in which we use Progress() followed by Playback().
I've added some code to chan_pjsip.c which seems to work. I've opened a ticket here for two reasons: one: the "hack" could be usefull for someone else and second: I wanted to ask whether it is safe to do it the way I solved this or whether I'd need to lock channel->session->channel or something else first?
--- asterisk-13.18.0/channels/chan_pjsip.c 2017-10-30 16:33:07.000000000 +0100
+++ asterisk-13.18.0_mod/channels/chan_pjsip.c 2017-11-29 12:02:55.000000000 +0100
@@ -1335,6 +1335,15 @@
res = -1;
} else {
response_code = 180;
+
+ const char *ref_by_val;
+ ref_by_val = pbx_builtin_getvar_helper(channel->session->channel, "PROGRESSINBAND");
+ if (!ast_strlen_zero(ref_by_val)) {
+ if (ref_by_val[0]=='1') {
+ response_code = 183;
+ res = -1;
+ }
+ }
}
} else {
res = -1;
--
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