[asterisk-users] internal sounds of asterisk / freePBX

[shadowym]

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