diff options
| author | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
| commit | ac091d41844a9a4a0f7d539164bcd154351b6da7 (patch) | |
| tree | 84ec4d1350b4e6d1d1498c4ceabd2b5484f3947d /channels/pjsip | |
| parent | 285be15aaf0469055d3392ecd73eb24395e49059 (diff) | |
chan_pjsip: Add support for passing hold and unhold requests through.
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/pjsip')
| -rw-r--r-- | channels/pjsip/dialplan_functions.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c index 6c0aff30b..6cc88017a 100644 --- a/channels/pjsip/dialplan_functions.c +++ b/channels/pjsip/dialplan_functions.c @@ -434,7 +434,7 @@ static int channel_read_rtp(struct ast_channel *chan, const char *type, const ch } else if (!strcmp(type, "secure")) { snprintf(buf, buflen, "%d", media->srtp ? 1 : 0); } else if (!strcmp(type, "hold")) { - snprintf(buf, buflen, "%d", media->held ? 1 : 0); + snprintf(buf, buflen, "%d", media->remotely_held ? 1 : 0); } else { ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type); return -1; |