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<DIV><FONT face=Arial size=2>It seems that the "other end" is having an issue
authenticating you. I have seen lots of switches act up with asterisk if you
don't tweak you settings in sip.conf just right. Do a SIP debug and have a look
at te INVITE request.</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=rurounitsukasa@gmail.com
href="mailto:rurounitsukasa@gmail.com">Ondřej Polívka</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Saturday, August 18, 2007 3:25
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [asterisk-users] incoming calls
in SIP</DIV>
<DIV><BR></DIV>
<DIV class=moz-text-flowed lang=x-unicode
style="FONT-SIZE: 12px; FONT-FAMILY: -moz-fixed">Hi, when I try to call in it
tells me: NOTICE[11664]: chan_sip.c:10637 handle_request_invite: Failed to
authenticate user "585415198" <A class=moz-txt-link-rfc2396E
href="mailto:585415198@82.208.46.240>"><sip:585415198@82.208.46.240></A>;tag=as18abefe8
<BR>Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04.
Outcoming and internal calls functions well. Thanks <BR><BR>sip.conf:
<BR>[general] <BR>callevents=yes <BR><BR>register => <A
class=moz-txt-link-abbreviated
href="mailto:username:password@sip.mujtelefon.cz/username">username:password@sip.mujtelefon.cz/username</A>
<BR><BR>[100] <BR>callerid=100 <BR>secret=password <BR>type=friend
<BR>context=internal <BR>srvlookup=yes <BR>type=friend <BR>qualify=yes
<BR>nat=no <BR>host=dynamic <BR>canreinvite=no <BR>context=internal
<BR>call-limit=1 <BR><BR>[101] <BR>callerid=101 <BR>secret=password
<BR>type=friend <BR>context=internal <BR>srvlookup=yes <BR>type=friend
<BR>qualify=yes <BR>nat=no <BR>host=dynamic <BR>canreinvite=no
<BR>context=internal <BR>call-limit=1 <BR><BR>[TRUNK-587207103]
<BR>type=friend <BR>context=incoming <BR>username=username <BR>secret=password
<BR>host=<A href="http://sip.mujtelefon.cz">sip.mujtelefon.cz</A>
<BR>dtmfmode=info <BR>canreinvite=no <BR>qualify=no <BR>nat=yes
<BR>disallow=all <BR>allow=g729 <BR>allow=gsm <BR>allow=alaw
<BR><BR><BR><BR></DIV>
<P>
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