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Summary:ASTERISK-11833: Single duplex channels in 1.4.19
Reporter:Luke-Jr (luke-jr)Labels:
Date Opened:2008-04-11 19:51:23Date Closed:2011-06-07 14:03:20
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/General
Versions:Frequency of
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Description:After upgrading to 1.4.19, *almost* all my incoming calls are single duplex: I can hear the caller, but they cannot hear me. This is via both SIP and IAX2. Outbound calls (originated by PAP2s) work fine. Downgrading to Asterisk 1.4.18.1 fixes the problem.
Comments:By: Luke-Jr (luke-jr) 2008-04-11 20:31:14

I've been examining the packet streams and thus far the only difference I can see is that 1.4.19 sets the marker bit on some audio RTP packets.
By: Luke-Jr (luke-jr) 2008-04-12 01:42:56

Bug origin identified:

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r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines
Changed paths:
  M /branches/1.4/channels/chan_sip.c
  M /branches/1.4/include/asterisk/sched.h

Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue ASTERISK-11574)
Reported by: slavon
Patches:
      sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)

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By: Luke-Jr (luke-jr) 2008-04-12 10:15:50

Reverting this revision in 1.4.19 does not, however, fix the problem there.
By: Luke-Jr (luke-jr) 2008-04-12 10:17:02

I'm lost, and this blocks upgrading 1.4 for me.
By: Joshua C. Colp (jcolp) 2008-04-14 09:38:37

1. Try latest 1.4 from SVN
2. Provide complete console output with debug enabled at the core level
3. If it's SIP then rtp debug is needed to see the frames going in and potentially not out.
4. Provide a core show channel <channel name> from each side
By: Jason Parker (jparker) 2008-05-01 14:44:28

Closing, due to lack of response/information.