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
