[asterisk-users] g729

[Jaswinder Singh]

Just read somewhere that you can use extension as g729 even in
mixmonitor so it will record g729 stream and later you can convert it
to mp3 or wav using sox . If this fails then try monitor application .


On 06/06/07, Jaswinder Singh <vicky.r at gmail.com> wrote:
> 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 <enunez at netoneint.com> 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/3756${EXTEN}@nextone,,wW)
> >
> >
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] 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 <enunez at netoneint.com> 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
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