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Summary:ASTERISK-14212: Dropping frame since I'm still dialing on DAHDi/1-1...
Reporter:Vadim Sherbakov (vinsik)Labels:
Date Opened:2009-05-27 05:04:02Date Closed:2009-09-28 12:29:12
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_dahdi
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 1.6-asterisk-dahdi-dialing.txt
( 1) 1.6-asterisk-dahdi-dialing-cli
Description:chan_dahdi seems to think that call is still in dialing state.
All though it is not. Call is answered on the other side.

I have tried callprogress=yes, (both sip.conf and chan_dahdi.conf)

And weird thing is when i press a DTMF key, channel gets established and
audio is working fine.

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

attachment: debug file, console output
Comments:By: Vadim Sherbakov (vinsik) 2009-05-29 02:05:58

I installed 1.4.23.1 with dahdi 2.2 , and it works.
By: Udo Schacht-Wiegand (udosw) 2009-06-10 05:36:29

Same problem is reproduceable with 1.4.25.
Strange: It only happens with some numbers I call via the Zap (DAHDI) channel, with others there is no problem.
As a workaround I now send a DTMF to the called party, and Audio is established immediately: Dial(Zap/.../.......,,D(0)).


By: Mark Clarke (mxc) 2009-06-27 02:36:34

I can confirm this issue. Incoming calls work fine. On answer everything works as expected. It is just for outgoing calls. When the callee answers asterisk set the state of the device to in use but it does not connect. There is no audio. In the case of DADHI extensions being called on DIDs, when the extension is answered you just get dailling tone played back. The other side hears silence. For sip phone silence both ways. Here is some log file

[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Requested indication 20 on channel DAHDI/1-1
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Requested indication 20 on channel DAHDI/2-1
[Jun 27 09:03:50] DEBUG[29812] devicestate.c: No provider found, checking channel drivers for DAHDI - 2
[Jun 27 09:03:50] DEBUG[29812] devicestate.c: Changing state for DAHDI/2 - state 2 (In use)
[Jun 27 09:03:50] DEBUG[29812] devicestate.c: No provider found, checking channel drivers for DAHDI - 1
[Jun 27 09:03:50] DEBUG[29812] devicestate.c: Changing state for DAHDI/1 - state 2 (In use)
[Jun 27 09:03:50] DEBUG[29816] app_queue.c: Device 'DAHDI/2' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Jun 27 09:03:50] DEBUG[29816] app_queue.c: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Dropping frame since I'm still dialing on DAHDI/2-1...
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Dropping frame since I'm still dialing on DAHDI/2-1...
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Dropping frame since I'm still dialing on DAHDI/2-1...
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Dropping frame since I'm still dialing on DAHDI/2-1...
[Jun 27 09:03:50] DEBUG[31410] chan_dahdi.c: Dropping frame since I'm still dialing on DAHDI/2-1...
By: destiny6628 (destiny6628) 2009-07-01 03:05:20

what is the setting for overlap dial in chan_dahdi.conf or zapata.conf.

i faced same problem few days back and setting overlapdial=no fixed my problem.

I was also getting voice on incoming calls but on outgoing calls after pressing dtmf only voice was through.

May be this helps.
By: Alec Davis (alecdavis) 2009-07-03 07:38:34

Possibly related to https://issues.asterisk.org/view.php?id=15420
By: Digium Subversion (svnbot) 2009-07-09 18:37:53

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 18:46:23

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 18:51:51

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 18:56:21

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
By: Leif Madsen (lmadsen) 2009-07-13 07:18:24

vinsik: please test latest Asterisk after revision 205728 and let us know if the commits have resolves this issue as well. Thanks!
By: Leif Madsen (lmadsen) 2009-07-20 14:49:17

I'm closing this as this was probably fixed by commit #205728.