> -- > www.ilovetovoip.com > > > On 2010-09-11 7:22 PM, "Paul Belanger" > > <paul.belan...@polybeacon.com> wrote: > > > > > > > > On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere <j...@sunfone.com> > > wrote: > > >> Sending to 123.123.12... > > > > > Either you changed the peer parameters or they did... > > > > > > > If he is not receiving any response, it is most likely a routing > > issue. > > > > -- > >
[un top posting] On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote: > Actually it is a very easy to understand and fix issue, but looking at > the code taking care of anonymous sip calls is the key. Those who post > third party GUI related issues should at least post the underlying > asterisk config or code here, so the asterisk part of the problem can > be fixed. > > > Zeeshan A Zakaria > > Its not that he isn't receiving a response - its that his peer debug statement isn't getting tripped because the peer hasn't authenticated. That's why I suggested he debug by IP rather than peer. Then what he will see is the SIP auth attempts and asterisk rejecting them, but in my experience not much is of value in seeing those packets - it doesn't point to *why* the connection is being rejected. The routing must be ok since allowing guest sip connections (the result of setting "accept anonymous" in FreePBX) allows the calls to come in fine. His problem is the peer authenticating. This of course has nothing to do with extensions.conf, as the dialplan is not involved. It is a SIP authentication problem, purely. There is no "relevant code" to post, and if you had ever looked into FreePBX's "relevant code" you would realize that it is actually fairly complex, and you would indeed have a difficult time debugging the flow. It *might* help if he posted his peer entry, but without seeing the other side that may not help much either. As Paul suggested first off, he should be in touch with his provider, whose tech support should be able to help him sort it out. I ran into a strange one EXACTLY like this just last week. We have a residential dial-tone customer with a Linksys SPA2102 (our standard device for this service). He had someone come out and replace his home router, and when he did he stopped authenticating. He has a fixed IP, so I enabled the debugging as I have mentioned twice now (by IP) and saw the attempts and rejections. After much hair pulling I *disabled* nat in his peer entry and it suddenly connected fine. This is bizarre, as our standard peer configuration works for 100% of the rest of our customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users