Hi James, I tried putting the Wait there but it is still the same too... Thanks alot for your help.
Pete On Mon, Mar 17, 2008 at 9:04 PM, James Texter III <[EMAIL PROTECTED]> wrote: > Try putting in a wait after you answer. It's possible the message is > playing before the RTP is setup. I would change your dialplan to be > > exten => 333,1,Answer() > exten => 333,n,Wait(1) > exten => 333,n,Playback(vm-goodbye) > exten => 333,n,Hangup() > > HTH, > > James > > On Mar 17, 2008, at 5:47 AM, Anselm Martin Hoffmeister wrote: > > > Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay: > >> Hi, > >> I am new to Asterisk and I am having a setup problem that I am trying > >> to resolved for the last couple days without any success. I am > >> pretty > >> much desperated on this issue and I don't know why. Can someone > >> please kindly help me to troubleshoot this? I can't hear any audio > >> from Asterisk when running Playback or VoiceMail tests. > > > > Dear Pete, > > > > my first idea would be that something with your codecs is borken > > (TM). I > > personally use a setup quite similar to yours, with the one visible > > difference that I also allow the "gsm" codec, owing to the fact that > > at > > least my home-recorded prompts are gsm only. I _guess_ asterisk > > could or > > should handle format conversion from audio files automagically, but > > for > > making sure, please try adding "gsm", at least for now. > > > > You might also want to setup the > > [sipclient] stanza in sip.conf such that "nat" is set to "no", > > although > > I do not see why that should break things. Especially as "Echo" works. > > > > The externip is set to your current external IP, right? (Knowing full > > well that some DSL lines get a new IP as often as 6 times a day, or > > as a > > P2P bandwidth countermeasure down to five minute intervals at certain > > restrictive providers once your "fair use" volume is used up). Again > > this should not be the culprit... > > > > Poking with a stick in the swamps, but perhaps hitting the bug :-P > > > > BR > > Anselm > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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