diff options
| author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-12 14:32:22 +0000 |
|---|---|---|
| committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-10-12 14:32:22 +0000 |
| commit | c361f63aac67b830b2726c864ec1c85cdcc67182 (patch) | |
| tree | 0c8be1a6c2698076b9950d526cac981cd4ecabbb /channels | |
| parent | c3397ff7638f41832193551e8f2d679213d7685a (diff) | |
Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.
(closes issue #16025)
Reported by: jamicque
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223654 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 3 |
1 files changed, 1 insertions, 2 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 98b05ae18..2ab5a6fd2 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5598,8 +5598,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame) ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); } } else if (p->t38.state == T38_ENABLED) { - change_t38_state(p, T38_DISABLED); - transmit_reinvite_with_sdp(p, FALSE, FALSE); + /* drop frame, can't sent VOICE frames while in T.38 mode */ } else { p->lastrtptx = time(NULL); res = ast_rtp_write(p->rtp, frame); |