Just wanted to update anyone interested in this issue. If I monitor a g729 SIP channel using ChanSpy, I am getting the same error as when I use MixMon.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Thursday, June 07, 2007 12:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] g729 Oddly enough the call was being recorded. In any case in case anyone is having the same problem, here is what did to get rid of the errors. I am now using Monitor instead of MixMonitor as Jaswinder suggested. Thanks exten => _1NXXNXXXXXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE RID:6}-${EXTEN}-${TIMESTAMP}-OUT) exten => _1NXXNXXXXXX,2,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Set(CDR(UserField)=${MONITOR_FILENAME}) exten => _1NXXNXXXXXX,4,Set(CALLERID(number)=14073844200) exten => _1NXXNXXXXXX,5,Monitor(${CALLFILENAME}.wav49||mb) exten => _1NXXNXXXXXX,6,Dial(SIP/[EMAIL PROTECTED]) -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wednesday, June 06, 2007 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 I think asterisk first converts audio stream to slin for recording to a wav file . Since you are using hardware g729 transcoder i think this is what is causing the problem . Is the calla actually being recorded ? I suggest that you use monitor application since it can directly record g729 audio stream and run some cron script with sox mixing the IN and OUT files in 1 file . On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote: > Yes > > This is my extensions.conf entry. > > exten => _1NXNXXXXXXX,1,Set(DYNAMIC_FEATURES=automon) > exten => > _1NXXNXXXXXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE > RID}-${EXTEN}-${TIMESTAMP}-OUT) > exten => > _1NXXNXXXXXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}- > OUT) > exten => _1NXXNXXXXXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP}) > exten => _1NXXNXXXXXX,5,Set(CDR(UserField)=${MONITOR_FILENAME}) > exten => _1NXXNXXXXXX,6,Set(CALLERID(number)=14073844200) > exten => _1NXXNXXXXXX,7,MixMonitor(${CALLFILENAME}.wav49) > exten => _1NXXNXXXXXX,8,Dial(SIP/[EMAIL PROTECTED],,wW) > > > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder > Singh > Sent: Wednesday, June 06, 2007 4:28 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] g729 > > Are you trying to record the conversation as well ? > > On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote: > > > > > > > > > > I installed a hardware g729 codec card in my asterisk, and I'm getting the > > following error when calling from a g729 sip extension to a SIP trunk also > > set to g729. The call goes through just fine, but these error messages > keep > > flying by until I disconnect the call. > > > > > > > > Any ideas? > > > > > > > > ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin > > failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > > > Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: > > Translation to slin failed, dropping frame for spies > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
