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 <[EMAIL PROTECTED]> wrote:
The codecs are correct, so you are having other type of problem -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>: > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Carlos Jerónimo _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- ------ Christopher T Aloi ------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
