CallWeaver is the new name for OpenPBX -----Original Message----- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>: > 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 > > > > _______________________________________________ > --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
