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<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2>My configs that I've reworked in the process of trying to
fix this SIP problem actually started from Freepbx.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2>I removed and reinstalled Asterisk last night, things seem
to be working smoother, I'll no by noon if the problem is fixed or
not.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2>Thanks for the help from everyone,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=655151013-10052007><FONT face=Arial
color=#0000ff size=2>Ken</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Deepak
Naidu<BR><B>Sent:</B> Wednesday, May 09, 2007 11:54 PM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> RE:
[asterisk-users] SIP Problems continue...<BR></FONT><BR></DIV>
<DIV></DIV>
<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
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