Summary:	ASTERISK-13921: So much silence inserted in recorded sound files by Manager Monitor
Reporter:	evandro (evandro)	Labels:
Date Opened:	2009-04-08 07:26:14	Date Closed:	2011-06-07 14:03:10
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Resources/res_monitor
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) 20090423_142923_6202_Anderson-Mesa_48_O_6405.mp3 ( 1) 20090423_142923_6202_O_6405_6202-in.wav ( 2) 20090423_142923_6202_O_6405_6202-out.wav ( 3) 20090423_142923_6405_I_6202_6405-in.wav ( 4) 20090423_142923_6405_I_6202_6405-out.wav ( 5) 20090423_142939_6412_Adenilson-SIP-2_46_I_6405.mp3 ( 6) 20090423_142939_6412_I_6405_6412-in.wav ( 7) 20090423_142939_6412_I_6405_6412-out.wav
Description:	How it happens: A receives a call from B. A select another line in his softphone, putting B on hold. A calls C, speak for a little time, and then transf C to B, ending his call. On B side of recorded sound file, is inserted silence while B was on hold, but this time is greater than length of hold. This cause a big delay on B side, making the conversation get out of sync. Any help? ****** STEPS TO REPRODUCE ****** B --> A A --> attended transfer --> C B on hold B --> C Hangup (I'm putting this here because the description confused me that I thought A called B, then A did the transfer. I'd have worded that a bit differently :))
Comments:	By: Leif Madsen (lmadsen) 2009-04-23 11:29:50 I'm unable to reproduce this issue. I've tried several ways of trying to make it happen on 1.4.23, but simply can't get it to happen. Please reopen should you still have this issue. You'll need to attach a sample audio file that is showing the symptoms, along with the relative parts of your dialplan and sip.conf files. Also be clear what technologies you are using (all SIP? DAHDI / SIP / IAX2?), and how long you are waiting on hold, and how long A is talking to C. By: evandro (evandro) 2009-04-23 14:00:59 In this case, the phone 6202(dahdi/7) calls 6405(sip/6405), who transfer him to 6412(sip/6412). I attached the record files generated by manager monitor through recording dahdi/7 and sip/6412, and also already attached the mixed files. When mix dahdi/7 files (20090423_142923_6202_O_6405_6202-in.wav and 20090423_142923_6202_O_6405_6202-out.wav) the result (20090423_142923_6202_Anderson-Mesa_48_O_6405.mp3) is ok, but when mix sip/6412 files (20090423_142939_6412_I_6405_6412-out.wav and 20090423_142939_6412_I_6405_6412-in.wav), is inserted the too long silence that I mentioned. In this case, dahdi/7 was on hold for nearly 17 seconds, and sip/6405 talking with sip/6412 for 7 seconds. my dialplan configs: [general] static=yes writeprotect=no autofallthrough=no clearglobalvars=no the context where the calls was: [Saida] exten => _XXXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,1) exten => _XXXXXXXX,n,Goto(Ocupado,s,1) exten => _0XXXX[1-6]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,3) exten => _0XXXX[1-6]XXXXXXX,n,Goto(Ocupado,s,1) exten => _0XXXX[7-9]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,4) exten => _0XXXX[7-9]XXXXXXX,n,Goto(Ocupado,s,1) exten => _9948995[3467],1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,6) exten => _9948995[3467],n,Goto(Ocupado,s,1) exten => _99499972,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,7) exten => _99499972,n,Goto(Ocupado,s,1) exten => _99604212,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,8) exten => _99604212,n,Goto(Ocupado,s,1) exten => _99630124,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,9) exten => _99630124,n,Goto(Ocupado,s,1) exten => _99691311,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,10) exten => _99691311,n,Goto(Ocupado,s,1) exten => _0800.,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,12) exten => _0800.,n,Goto(Ocupado,s,1) exten => _3[23]54XXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,23) exten => _3[23]54XXXX,n,Goto(Ocupado,s,1) exten => _0XX[1-6]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,24) exten => _0XX[1-6]XXXXXXX,n,Goto(Ocupado,s,1) exten => _XXXX[1-6]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,25) exten => _XXXX[1-6]XXXXXXX,n,Goto(Ocupado,s,1) exten => _XX[1-6]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,26) exten => _XX[1-6]XXXXXXX,n,Goto(Ocupado,s,1) exten => _XXXX[7-9]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,27) exten => _XXXX[7-9]XXXXXXX,n,Goto(Ocupado,s,1) exten => _0XX[7-9]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,28) exten => _0XX[7-9]XXXXXXX,n,Goto(Ocupado,s,1) exten => _XX[7-9]XXXXXXX,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,29) exten => _XX[7-9]XXXXXXX,n,Goto(Ocupado,s,1) ;exten => _X.,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,30) exten => _X.,1,Set(calltime = => ${STRFTIME(${EPOCH},,%T)}) exten => _X.,n,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,30) ;exten => _X.,1,Noop( SIPTRANSFER = ${SIPTRANSFER} ) ;exten => _X.,n,Noop( BLINDTRANSFER = ${BLINDTRANSFER} ) ;exten => _X.,n,Noop( CALLERID = ${CALLERID(num)} ) ;exten => _X.,n,Noop( TRANSFERERNAME = ${TRANSFERERNAME} ) ;exten => _X.,n,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,30) exten => _X.,n,Goto(Ocupado,s,1) exten => _*144,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,1,31) exten => _*144,n,Goto(Ocupado,s,1) exten => _*5,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,3) exten => _*5,n,Goto(Ocupado,s,1) exten => _*5.,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,3) exten => _*5.,n,Goto(Ocupado,s,1) exten => _*6[1234].,1,AGI(/dg/intelix/configurador/agi-bin/ligacaoSaida.agi,4) exten => _*6[1234].,n,Goto(Ocupado,s,1) my sip configs: [general] context=default                 allowoverlap=no                 bindport=5060 bindaddr=0.0.0.0 srvlookup=yes language=pt_BR [6405] callerid="Evandro-Sip"<6405> context=Saida type=peer secret=Evandro host=dynamic permit=0.0.0.0/0.0.0.0 nat=no call-limit=10 qualify=yes allowtransfer=yes [6412] callerid="Adenilson-Sip2"<6412> context=Saida type=peer secret=Adenilson2 host=dynamic permit=0.0.0.0/0.0.0.0 nat=no call-limit=10 qualify=yes allowtransfer=yes By: David Brooks (dbrooks) 2009-05-28 14:17:26 evandro, in the description of the bug, you say B calls A, A puts B on hold and calls C, then transfers C to B. But in the note you posted, you say B calls A, but then A transfers B to C. Is this error happening with all transfers? Or do you mean that the error only happens during the recording? By: evandro (evandro) 2009-05-28 14:33:58 dbrooks, I had make a mistake in the description of the bug. The error happens as decribed in the note, and only in recording. During a call is all normal. It's just happens when the transferer is a softphone and made transfer by using two lines of phone. When using feature transfer resource or blind transfer on his softphone, it works well. By: Leif Madsen (lmadsen) 2009-05-28 16:00:30 Do you have the ability to try the latest 1.4 branch? A few patches have gone in recently to help with the monitor and chanspy applications to keep things in sync. Please verify you're still getting this issue with the latest 1.4 branch: svn co http://svn.asterisk.org/svn/asterisk/branches/1.4 Thanks! By: Leif Madsen (lmadsen) 2009-07-13 10:08:35 Closed due to lack of response. Please retest with the latest Asterisk from SVN, and if you still have an issue, please open a new issue. Thanks!