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 <[EMAIL PROTECTED]> wrote:
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
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