<div>A small way to make little easy, I dont know it people are ok to that, try integrating freepbx & asterisk so you know what the sip configs should look like when things are all well.</div> <div> </div> <div>Things might stop working if there is a bug or change in configs.</div> <div> </div> <div>--</div> <div>Deepak<BR><BR><B><I>Ken Williams <ken@intermountainelectronics.com></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">I mean that SIP phones cannot answer incoming calls or make outgoing<BR>calls. When a call comes in on ZAP, it actually rings all the phones<BR>like normal, but when you try to answer no one is there. In addition,<BR>when you try to dial out you eventually get a message on the phones<BR>saying unable to communicate with the server. So there is some traffic<BR>still traveling on the SIP channel (the server's dialing extensions from<BR>an incoming ZAP
call) but no further communication...almost as if it's a<BR>one way street of communication. The server can send data out on SIP<BR>but isn't receiving any.<BR><BR>As for your issue, we haven't really had that (thankfully), so I don't<BR>think you're heading down the horrible spot we're in right now.<BR><BR>Tonight I'm going to remove all aspects of Asterisk and reinstall fresh,<BR>if that fails I'll format & reinstall the entire box. <BR><BR>-----Original Message-----<BR>From: asterisk-users-bounces@lists.digium.com<BR>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam<BR>Moffett<BR>Sent: Wednesday, May 09, 2007 12:08 PM<BR>To: Asterisk Users Mailing List - Non-Commercial Discussion<BR>Subject: Re: [asterisk-users] SIP Problems continue...<BR><BR>I also get the mysterious SIP INVITE channels.<BR>10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init:<BR>INVITE<BR><BR>And I also am running 1.4.4 on CentOS4. Is that a pattern or
just<BR>coincidence?<BR><BR><BR><BR>The other symptom you mention is this<BR>"...the SIP phones couldn't communicate with the server, though there <BR>was no error message on the server and everything appeared fine on the <BR>server."<BR><BR>Do you mean no calls in or out until you reboot? I don't have that <BR>thankfully, but I do have a guy telling me that incoming audio just goes<BR><BR>away for a few seconds at a time. He says also that it sometimes goes <BR>away for long enough time that he was mistaking it for a dropped call. <BR>But if he waits long enough it pretty generally always comes back. I <BR>have consistent solid network performance from the asterisk server to <BR>the ATA (and believe me, I've looked very hard for a network problem), <BR>and I don't know what to look at next.<BR><BR>Incidentally, the guy hasn't called me since I rebooted last week. Is <BR>this similar to how your situation started?<BR><BR><BR><BR>*********************************<BR>Adam
Moffett<BR>Plexicomm, LLC<BR>adam@plexicomm.net<BR>ph: 866-759-4678x104<BR>*********************************<BR><BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR><p> 
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