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Summary:ASTERISK-14368: [patch] no audio with SIP call to ISDN PRI, if neither Progress or Proceeding are received.
Reporter:Alec Davis (alecdavis)Labels:
Date Opened:2009-06-24 08:01:46Date Closed:2009-09-18 09:49:56
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_dahdi
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 1-6-1-nonworking-4888.txt
( 1) bug15389_noaudio.diff.txt
( 2) bug15389_noaudio.diff2.txt
( 3) working-4888.txt
Description:After an upgrade from Asterisk SVN-r178446 to Asterisk 1.6.1-Branch.

Calling from SIP a phone to our ISDN PABX there is now no audio in either direction.
Calling from the PABX into an Asterisk SIP phone audio is fine.

Audio on outbound call works fine on Asterisk 1.6.1.0-rc3 Tag,
but is broken on Asterisk 1.6.1.0-rc4 Tag.

The original patch (noted below) needs further work, to ensure audio path is up when call is 'answered', may be earlier, when 'ringing' is received.

****** ADDITIONAL INFORMATION ******

Using Asterisk 1.6.1 Branch and reverting the patches from https://issues.asterisk.org/view.php?id=13034 fixes the no audio.

Dahdi version: DAHDI Version: 2.2.0 Echo Canceller: MG2
Libpri version: libpri version: SVN-branch-1.4-r879

Console output below: Main intention is to show simple dial, and the fact that only 'ringing' and 'answered' are seen.

License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
 == Parsing '/etc/asterisk/asterisk.conf':   == Found
 == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk SVN-branch-1.6.1-r202764M currently running on astrid (pid                                                 = 20379)
Verbosity is at least 4
 == Using SIP RTP CoS mark 5
   -- Executing [4888@trusted:1] Dial("SIP/8530-09e482e0", "DAHDI/G0/4888") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called G0/4888
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/31-1 is ringing
   -- DAHDI/31-1 answered SIP/8530-09e482e0
astrid*CLI>
Comments:By: Alec Davis (alecdavis) 2009-06-24 08:14:35

uploaded:
1-6-1-nonworking-4888.txt: PRI debug capture of call to 4888 with no audio present, with Asterisk SVN-branch-1.6.1-r202764.

working-4888.txt: PRI debug capture of call to 4888 with audio present, this was with downgraded asterisk SVN-r178446

There's no diffence at the ISDN level.

What I forgot to add in the main description, was that if you pressed any button on the phone that created DTMF while there should have been audio present, that audio was then present for the remainder of the call.
By: Alec Davis (alecdavis) 2009-06-25 07:25:40

please remove bug15389_noaudio.diff.txt
after cleanup I removed dialing=0, and didn't test before submitting.
Note to me: Always test after cleanup, before submitting patch!!!!

uploaded bug15389_noaudio.diff2.txt:
if ISDN Proceeding and Progress Indicators are not seen, and the call is answered the 'dialing' flag is left set, preventing an audio path setup.

This essentially 1 line patch, sets 'dialing' flag to 0, when PRI_EVENT_ANSWER is fired.

=======================================
Console output below:
- 1st call to PABX without PROGRESS or PROCEEDING indicators
- 2nd call to PSTN with both PROCEEDING and PROGRESS.

Both calls were sucessful, both with Audio path.
Previously only the 2nd call below would have had an audio path.
=======================================

 == Using SIP RTP CoS mark 5
   -- Executing [4888@trusted:1] Dial("SIP/8630-08cbefe8", "DAHDI/G0/4888") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called G0/4888
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/31-1 is ringing
   -- DAHDI/31-1 answered SIP/8630-08cbefe8
   -- Hungup 'DAHDI/31-1'
 == Spawn extension (trusted, 4888, 1) exited non-zero on 'SIP/8630-08cbefe8'
 == Using SIP RTP CoS mark 5


   -- Executing [1083210@trusted:1] Dial("SIP/8630-08cc5a58", "DAHDI/G0/083210,,r") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called G0/083210
   -- DAHDI/31-1 is proceeding passing it to SIP/8630-08cc5a58
   -- DAHDI/31-1 is making progress passing it to SIP/8630-08cc5a58
   -- DAHDI/31-1 is making progress passing it to SIP/8630-08cc5a58
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/31-1 answered SIP/8630-08cc5a58
   -- Hungup 'DAHDI/31-1'
By: Alec Davis (alecdavis) 2009-06-25 07:31:45

The testing done was with SVN-branch-1.6.2-r203077M although this bug was initially set as 1.6.1

Looking at Trunk it also applies, but is untested.
By: Loris Santamaria (loris) 2009-07-03 07:30:38

We're seeing the same issue since upgrade from 1.4.23.2 to 1.4.25.1. Seems to be the same problem as issue 15205
By: Alec Davis (alecdavis) 2009-07-03 07:44:37

For 1.4.25.1 refer to https://issues.asterisk.org/view.php?id=15420 which is ready for review.
By: Alec Davis (alecdavis) 2009-07-04 05:53:56

please remove bug15389_noaudio.diff.txt
By: Alec Davis (alecdavis) 2009-07-06 18:26:41

please remove bug15389_noaudio.diff.txt

testing with patch bug15389_noaudio.diff2.txt:

From asterisk SIP phone dialled with and without 'w' option to PABX, where PROCEEDING and PROGRESS are not received, only ALERTING then CONNECT, dtmf was successfully heard at the remote handset if required, then audio passed through.

   -- Called r0/4866w4866
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/3-1 answered SIP/GXP0001-084b1340
   -- Channel 0/3, span 1 got hangup request, cause 16
   -- Hungup 'DAHDI/3-1'

   -- Called r0/4866
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/4-1 is ringing
   -- DAHDI/4-1 answered SIP/GXP0001-084a7c20
   -- Channel 0/4, span 1 got hangup request, cause 16
   -- Hungup 'DAHDI/4-1'

From asterisk SIP phone dialled without 'w' option, to TELCO where PROCEEDING and PROGRESS are received, audio passed through on answer.

   -- Called r0/083210
   -- DAHDI/5-1 is proceeding passing it to SIP/GXP0001-084a7c20
   -- DAHDI/5-1 is making progress passing it to SIP/GXP0001-084a7c20
   -- DAHDI/5-1 is making progress passing it to SIP/GXP0001-084a7c20
!! Unknown IE 50 (cs5, Unknown Information Element)
   -- DAHDI/5-1 answered SIP/GXP0001-084a7c20
   -- Hungup 'DAHDI/5-1'
By: Digium Subversion (svnbot) 2009-07-09 18:38:00

Repository: asterisk
Revision: 205728

U   branches/1.4/channels/chan_dahdi.c

------------------------------------------------------------------------
r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines

No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.

Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
(i.e. When libpri generates the event PRI_EVENT_ANSWER.)

(closes issue ASTERISK-14397)
Reported by: scottbmilne
Patches:
     bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis

(closes issue ASTERISK-14393)
Reported by: avinoash

(closes issue ASTERISK-14368)
Reported by: alecdavis

This patch should also fix the following issue:
(issue ASTERISK-14212)
Reported by: vinsik

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=205728
By: Digium Subversion (svnbot) 2009-07-09 18:46:27

Repository: asterisk
Revision: 205729

U   branches/1.6.0/channels/chan_dahdi.c

------------------------------------------------------------------------
r205729 | rmudgett | 2009-07-09 18:46:22 -0500 (Thu, 09 Jul 2009) | 28 lines

Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
 
 No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
 
 Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
 (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
 
 (closes issue ASTERISK-14397)
 Reported by: scottbmilne
 Patches:
       bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
 Tested by: scottbmilne, alecdavis
 
 (closes issue ASTERISK-14393)
 Reported by: avinoash
 
 (closes issue ASTERISK-14368)
 Reported by: alecdavis
 
 This patch should also fix the following issue:
 (issue ASTERISK-14212)
 Reported by: vinsik
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=205729
By: Digium Subversion (svnbot) 2009-07-09 18:51:55

Repository: asterisk
Revision: 205730

U   branches/1.6.1/channels/chan_dahdi.c

------------------------------------------------------------------------
r205730 | rmudgett | 2009-07-09 18:51:50 -0500 (Thu, 09 Jul 2009) | 28 lines

Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
 
 No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
 
 Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
 (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
 
 (closes issue ASTERISK-14397)
 Reported by: scottbmilne
 Patches:
       bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
 Tested by: scottbmilne, alecdavis
 
 (closes issue ASTERISK-14393)
 Reported by: avinoash
 
 (closes issue ASTERISK-14368)
 Reported by: alecdavis
 
 This patch should also fix the following issue:
 (issue ASTERISK-14212)
 Reported by: vinsik
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=205730
By: Digium Subversion (svnbot) 2009-07-09 18:56:26

Repository: asterisk
Revision: 205731

U   branches/1.6.2/channels/chan_dahdi.c

------------------------------------------------------------------------
r205731 | rmudgett | 2009-07-09 18:56:21 -0500 (Thu, 09 Jul 2009) | 28 lines

Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
 
 No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
 
 Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
 (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
 
 (closes issue ASTERISK-14397)
 Reported by: scottbmilne
 Patches:
       bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
 Tested by: scottbmilne, alecdavis
 
 (closes issue ASTERISK-14393)
 Reported by: avinoash
 
 (closes issue ASTERISK-14368)
 Reported by: alecdavis
 
 This patch should also fix the following issue:
 (issue ASTERISK-14212)
 Reported by: vinsik
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=205731