SIP debug output please. Thanks, Steve Totaro
On Mon, Mar 17, 2008 at 7:17 AM, Pete Kay <[EMAIL PROTECTED]> wrote: > 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 > > > > > _______________________________________________ > -- 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
