[asterisk-users] internal sounds of asterisk / freePBX

Carlos Jerónimo carjer at gmail.com
Mon Apr 23 20:09:26 MST 2007


Hi chris. The result it is the same, no sound.

-- Executing Answer("SIP/7010-081f6f68", "") in new stack
    -- Executing Playback("SIP/7010-081f6f68", "beep") in new stack
    -- Playing 'beep' (language 'en')

more sugestion?


2007/4/18, Christopher Aloi <chris.aloi at gmail.com>:
> Try getting rid of all those macros etc.. so you can see what's going
> on, something simple like:
>
> exten => 500,1,Answer()
> exten => 500,n,Playback(beep)
> exten => 500,n,Hangup()
>
> Then dial 500 from your soft phone and see what happens.
>
>
>
> On 4/17/07, EWV2 <mail at directlink.net.mx> wrote:
> > The codecs are correct, so you are having other type of problem
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Carlos
> > Jerónimo
> > Sent: Tuesday, April 17, 2007 5:10 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX
> >
> > HI, my sip.conf /codecs
> >
> > disallow=all
> > allow=ulaw
> > allow=alaw
> >
> > this codcs is correct?
> > thanks
> >
> >
> >
> > 2007/4/17, EWV2 <mail at directlink.net.mx>:
> > > It sounds like a codec problem.
> > >
> > > What codec are you using?
> > >
> > > If you are using g723.1 or g729 passthru you will not be able to hear
> > > nothing
> > >
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Carlos
> > > Jerónimo
> > > Sent: Tuesday, April 17, 2007 4:30 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [asterisk-users] internal sounds of asterisk / freePBX
> > >
> > > Sorry but i can't register in the freepbx forum, so this is my
> > > solutons for resolve my trouble.
> > >
> > > HI, my problem is with internal sounds of asterisk.
> > > for example when calling voicemail, no system recordings are being
> > > played back. However, when running asterisk
> > > in a debug mode, i see the call coming through to the system and the
> > > system playing back the wav files promptly.
> > >  However, no sound comes through. I have verified that the sounds are
> > > in the correct location and that
> > > asterisk:asterisk has access to all files, is music on hold works, but
> > > other than that no system recordings are audible.
> > >
> > > But this isn't just voicemail. It's every system recording. Such as
> > > the feature code *60 to
> > > play the current time. It shows the call connected and it shows to be
> > > playing the wav file, but nothing
> > > coming out of the speaker of the phone....didn't just try with one phone
> > > either
> > >
> > > In other words, asterisk shows it's all working well. my logs:
> > >
> > > == Spawn extension (macro-systemrecording, h, 1) exited non-zero on
> > > 'SIP/7010-081d7288'
> > >     -- Executing Macro("SIP/7010-0819b350", "user-callerid|") in new stack
> > >     -- Executing NoOp("SIP/7010-0819b350", "user-callerid: device
> > > 7010") in new stack
> > >     -- Executing GotoIf("SIP/7010-0819b350", "0?report") in new stack
> > >     -- Executing GotoIf("SIP/7010-0819b350", "0?start") in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "REALCALLERIDNUM=7010") in new
> > > stack
> > >     -- Executing NoOp("SIP/7010-0819b350", "REALCALLERIDNUM is 7010")
> > > in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "AMPUSER=7010") in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "AMPUSERCIDNAME=Portaria")
> > > in new stack
> > >     -- Executing GotoIf("SIP/7010-0819b350", "0?report") in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "CALLERID(all)=Portaria
> > > <7010>") in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "REALCALLERIDNUM=7010") in new
> > > stack
> > >     -- Executing NoOp("SIP/7010-0819b350", "TTL:  ARG1: ") in new stack
> > >     -- Executing GotoIf("SIP/7010-0819b350", "0?continue") in new stack
> > >     -- Executing Set("SIP/7010-0819b350", "_TTL=64") in new stack
> > >     -- Executing GotoIf("SIP/7010-0819b350", "1?continue") in new stack
> > >     -- Goto (macro-user-callerid,s,21)
> > >     -- Executing NoOp("SIP/7010-0819b350", "Using CallerID "Portaria"
> > > <7010>") in new stack
> > >     -- Executing Wait("SIP/7010-0819b350", "2") in new stack
> > >     -- Executing Macro("SIP/7010-0819b350",
> > > "systemrecording|dorecord") in new stack
> > >     -- Executing Goto("SIP/7010-0819b350", "dorecord|1") in new stack
> > >     -- Goto (macro-systemrecording,dorecord,1)
> > >     -- Executing Record("SIP/7010-0819b350",
> > > "/tmp/7010-ivrrecording:wav") in new stack
> > >     -- Playing 'beep' (language 'en')
> > >
> > > Really at a stand still until I can get this resolved so any thoughts
> > > are much appreciated.
> > >
> > >
> > > --
> > > Carlos Jerónimo
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> > --
> > Carlos Jerónimo
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>
> --
> ------
> Christopher T Aloi
> ------
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-- 
Carlos Jerónimo


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