-----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mark Wiater > Sent: Tuesday, February 03, 2009 2:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls > > Dropped in Voicemail > > Wouldn't this suggest that either Asterisk couldn't open the port, > or opened it and then closed it? Or I suppose that perhaps the phone > and asterisk didn't negotiate the port properly?
Since the RTP packets had up until that point been flowing in both directions, I'm guessing it's the middle (opened it and then closed it); no change in ports in the log, and this would seem to correspond roughly to the point at which Asterisk "gives up" on retrying the packet > In your original post, I thought I read that you could reproduce > this issue by increasing load on the asterisk server. What does the > caller experience in the first 20 seconds when a call to voicemail > is going to fail? Just ringing? I guess I should have been a little bit more clear on that one: Asterisk always answers calls to Voicemail more or less instantaneously. Reproducing it has been very hit-or-miss with no real correlation to network activity, call volume, time of day, day of week, phase of moon, etc. However, more often than not you can force the issue to manifest itself by by making several "rapid-fire" calls to voicemail from the same phone. The first 20 seconds of a call that ultimately fails is indistinguishable from a call that doesn't fail - Comedian Mail answers, accepts DTMF input from users, starts playing back messages, etc., and then all of a sudden at about the 20 second mark the audio dies. The phone still thinks that it's in a call but no more audio gets sent to the phone, DTMF input is ignored, etc. > Any chance there's anything in Asterisk's or the OSes logs about > some failure of the network stack? What OS is this? This is Ubuntu Server 8.1.0; as far as logs go, please excuse me -- my background is primarily Windows so I'm not sure where I would find those (but I'll Google that now ;) ) Also, an interesting side note: While I don't want to call the issue "fixed" yet based on how intermittent it has been, since I noticed that Asterisk indicated "Ringing" to the phone on SIP-to-SIP calls (that have never failed) I added: Voicemail,2,Ringing() Voicemail,3,Wait(1) To the Voicemail extension in the dialplan (I literally use "Voicemail" as the extension that the Messages button on the phone dials), and so far I have not been able to reproduce my issue... I don't like it because the user hears a ring cycle before the VM attendant answers, but if it keeps them from being bounced out of VM in the middle of listening to a message, I can live with it. Anyone with more knowledge of the inner workings of things want to tell me if I should or shouldn't be surprised if the issue reappears with this in place? Thanks again for the help! Lincoln _______________________________________________ -- 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
