Summary:ASTERISK-10391: ChannelRedirect non-working
Reporter:Dmitry Andrianov (dimas)Labels:
Date Opened:2007-09-26 10:22:58Date Closed:2011-06-07 14:03:21
Versions:Frequency of
Description:1. softphone 1011 dials 123
2. SIP phone 1010 starts ringing
3. 1010 answers, the conversation between 1010 and 1011 is perfect
4. 1011 activates bt2 feature macro by pressing *7
5. 1011 hangs up
6. "you are the only first person blah-blah-blah" is played for 1010
7. The problem is:
  *) if 1010 is a hardphone with VAD, as soon as I start producing some sound (and phone starts transmitting), I hear beeeeeeeeeeep in the 1010 headset
  *) if 1010 is a softphone without VAD and it transmits all the time, I hear constant beep. In fact I can not even hear "you are the only first person..." because of that beep.


bt2feat => *7,self/caller,Macro,bt2

context bt2context {
       _XXXX => {
               MeetMe(${EXTEN}, pdMX);

macro bt2() {


       123 => {

The console output below if for case when 1010 is Cisco hardphone (VAD/CNG). When it is a softphone the only difference if absence of CNG related notice...

   -- Executing [123@ael-default:1] Answer("SIP/1011-09dc4430", "") in new stack
   -- Executing [123@ael-default:2] Set("SIP/1011-09dc4430", "DYNAMIC_FEATURES=bt2feat") in new stack
   -- Executing [123@ael-default:3] Dial("SIP/1011-09dc4430", "SIP/1010") in new stack
   -- Called 1010
   -- SIP/1010-09dd56b0 is ringing
   -- SIP/1010-09dd56b0 answered SIP/1011-09dc4430
   -- Packet2Packet bridging SIP/1011-09dc4430 and SIP/1010-09dd56b0
   -- Packet2Packet bridging SIP/1011-09dc4430 and SIP/1010-09dd56b0
   --  Feature Found: bt2feat exten: bt2feat
   -- Executing [s@macro-bt2:1] ChannelRedirect("SIP/1011-09dc4430", "SIP/1010-09dd56b0|bt2context|6666|1") in new stack
 == Spawn extension (ael-default, 123, 3) exited non-zero on 'SIP/1011-09dc4430'
   -- Executing [6666@bt2context:1] MeetMe("SIP/1010-09dd56b0", "6666| pdMX") in new stack
   -- Created MeetMe conference 1015 for conference '6666'
   -- <SIP/1010-09dd56b0> Playing 'conf-onlyperson' (language 'en')
   -- Started music on hold, class 'default', on SIP/1010-09dd56b0
[Sep 26 19:37:27] NOTICE[18534]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
Comments:By: Digium Subversion (svnbot) 2007-10-15 11:01:22

Repository: asterisk
Revision: 85559

U   branches/1.4/main/rtp.c

r85559 | file | 2007-10-15 11:01:21 -0500 (Mon, 15 Oct 2007) | 4 lines

Bring both DTMF begin and end frames up through to the core for DTMF feature handling.
(closes issue ASTERISK-10391)
Reported by: dimas


By: Digium Subversion (svnbot) 2007-10-15 11:02:59

Repository: asterisk
Revision: 85560

_U  trunk/
U   trunk/main/rtp.c

r85560 | file | 2007-10-15 11:02:59 -0500 (Mon, 15 Oct 2007) | 12 lines

Merged revisions 85559 via svnmerge from

r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 lines

Bring both DTMF begin and end frames up through to the core for DTMF feature handling.
(closes issue ASTERISK-10391)
Reported by: dimas



By: Dmitry Andrianov (dimas) 2007-10-16 05:02:29

It looks like with the same configuration *7 does not activate feature at all anymore.

By: Joshua C. Colp (jcolp) 2007-10-16 08:43:44

Please provide complete console output with DTMF logging enabled in logger.conf as well as debug.

By: Dmitry Andrianov (dimas) 2007-10-16 09:10:28

in a SIP-SIP call (dtmfmode rfc2833), NO messages (neither debug nor DTMF) at all appear on console when I press buttons. When Zap is involved I see alot of DTMF messages when pressing buttons but with Zap channel I do NOT have these problems - DTMF features work just fine.

In 85559 you replaced several AST_FRAME_DTMF with AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END. Can it be that in some cases AST_FRAME_DTMF also needed to be in the "if" condition (like the one which decides to ast_write(other, fr))? (Sorry I do not really understand your code so my question can be just weird)

By: Joshua C. Colp (jcolp) 2007-10-16 09:13:56

AST_FRAME_DTMF_END is AST_FRAME_DTMF... as for seeing no DTMF information your issue is elsewhere, can you please attach a sip debug and rtp debug?

By: Joshua C. Colp (jcolp) 2007-11-12 08:30:13.000-0600

It's been a month without the required SIP/RTP info, please reopen when you have it.

By: Dmitry Andrianov (dimas) 2007-11-12 20:07:05.000-0600

Actually there was IRC discussion with you regarding the issue. And you discovered that everything was due to reinvites enabled - no DTMF were passing through Asterisk. So technically my problem is solved - it was configuration issue.

However you mentioned something like the fact that Asterisk did reinvite in this case or Asterisk did not warn me about consequences is the bug itself. So if you changed your mind - feel free to close it. Otherwise please try to remember the situation because I probably remember your words not exactly correct.

By: jmls (jmls) 2008-02-06 04:08:35.000-0600

file, can we close this ?

By: Joshua C. Colp (jcolp) 2008-02-06 09:32:21.000-0600

Yes, yes we can.