[asterisk-users] internal sounds of asterisk / freePBX

[EWV2]

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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