Hi, Thanks for pointing out. I checked the extenip and it is fine. The thing is that I have already configure gsm as one of the codec in the sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = others
register =>outraspace:[EMAIL PROTECTED]/outraspace nat=yes externip=58.251.75.333 localnet=192.168.1.0/255.255.255.0 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm qualify=yes Any other hints? On Mon, Mar 17, 2008 at 6:47 PM, Anselm Martin Hoffmeister < [EMAIL PROTECTED]> 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 >
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