Summary:	ASTERISK-13431: No voice (ringing tone) after call was diverted
Reporter:	radicalish (radicalish)	Labels:
Date Opened:	2009-01-22 09:39:39.000-0600	Date Closed:	2011-06-07 14:03:11
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) full.issue_14249.2
Description:	We have problem with one way voice in next scenario: 101 calls to IVR (number 1234) that have Answer() application and then call directed to 103, 103 doesn't want to answer and divert call (SIP 302 Moved Temporarily) to 102, 101 stops to hear ringing tones and we get in CLI bulk of messages: WARNING[5475] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8) when 102 answers audio come back to 101 I attached another log full.issue_14249.2
Comments:	By: radicalish (radicalish) 2009-01-22 10:13:14.000-0600 this issue related to bug 0014249 By: Digium Subversion (svnbot) 2009-01-23 13:07:18.000-0600 Repository: asterisk Revision: 170568 U   branches/1.4/apps/app_dial.c ------------------------------------------------------------------------ r170568 | file | 2009-01-23 13:07:18 -0600 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue ASTERISK-13431) Reported by: RadicAlish ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=170568 By: Digium Subversion (svnbot) 2009-01-23 13:08:51.000-0600 Repository: asterisk Revision: 170569 _U  trunk/ U   trunk/apps/app_dial.c ------------------------------------------------------------------------ r170569 | file | 2009-01-23 13:08:51 -0600 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........  r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines    When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.  (closes issue ASTERISK-13431)  Reported by: RadicAlish ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=170569 By: Digium Subversion (svnbot) 2009-01-23 13:09:34.000-0600 Repository: asterisk Revision: 170570 _U  branches/1.6.0/ U   branches/1.6.0/apps/app_dial.c ------------------------------------------------------------------------ r170570 | file | 2009-01-23 13:09:34 -0600 (Fri, 23 Jan 2009) | 18 lines Merged revisions 170569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................  r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines    Merged revisions 170568 via svnmerge from  https://origsvn.digium.com/svn/asterisk/branches/1.4    ........    r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines        When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.    (closes issue ASTERISK-13431)    Reported by: RadicAlish  ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=170570 By: Digium Subversion (svnbot) 2009-01-23 13:10:17.000-0600 Repository: asterisk Revision: 170571 _U  branches/1.6.1/ U   branches/1.6.1/apps/app_dial.c ------------------------------------------------------------------------ r170571 | file | 2009-01-23 13:10:17 -0600 (Fri, 23 Jan 2009) | 18 lines Merged revisions 170569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................  r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines    Merged revisions 170568 via svnmerge from  https://origsvn.digium.com/svn/asterisk/branches/1.4    ........    r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines        When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.    (closes issue ASTERISK-13431)    Reported by: RadicAlish  ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=170571 By: radicalish (radicalish) 2009-01-25 04:46:09.000-0600 svnbot: thanks for response I downloaded SVN-branch-1.4-r170836, but still have same problem with one difference: there are no WARNINGs. when 103 forward, 101 still loses voice (ringing tones) we are using temporarily 'r' option in Dial application as workaround By: Joshua C. Colp (jcolp) 2009-02-09 11:18:31.000-0600 I've tried numerous ways to reproduce this. Can you please attach an updated log WITHOUT sip debug but with core debug set to level 2? By: Leif Madsen (lmadsen) 2009-03-23 12:36:43 RadicAlish: Do you have the requested information, or an ETA when you could get the data? Or has the issue been resolved for you? I'll have to close this issue in about a week if we are unable to get a response. Thanks! Leif. By: Joshua C. Colp (jcolp) 2009-04-15 12:53:41 Suspended due to lack of response.