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/1.6.0@223653 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1 insertions, 2 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2f91cf523..da2f413f9 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5343,8 +5343,7 @@ 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);