[asterisk-users] internal sounds of asterisk / freePBX

Wed Apr 18 10:19:34 MST 2007

Did you have any E1/T1 cards in your server?

On 4/18/07, shadowym <shadowym at hotmail.com> wrote:
> CallWeaver is the new name for OpenPBX
>
> -----Original Message-----
> From: Carlos Jerónimo [mailto:carjer at gmail.com]
> Sent: Tuesday, April 17, 2007 3:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX
>
> i use xlite and kphone in a diferent pc's. i can phone well.
> the problem is internal asterisk sounds. I think i not use Call Weaver, what
> is call weaver, i search at google but i'm was confused.
>
> i hope more help's. thanks
>
>
>
>
> 2007/4/17, Andrew Joakimsen <joakimsen at gmail.com>:
> > If that's what your phone is setup. Are you even using a SIP phone?
> > What does the PEER context contain?
> >
> > Also, while Asterisk 1.2 and CALL WEAVER are basically the same
> > (besides that fact that CALL WEAVER is trying to fully support faxing
> > and Asterisk/Digium refuse to support correctly faxing) they do not
> > share sound files. So if you are indeed using CALL WEAVER and their
> > sounds you shouldn't be asking about that here.
> >
> > On 4/17/07, Carlos Jerónimo <carjer at gmail.com> wrote:
> > > 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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> > >
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> >
>
>
> --
> Carlos Jerónimo
>
>
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