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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>You shouldn’t “open text”
your password. Shouldn’t IP on Asterisk 2 be 1.2.3.4?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Shaun Wingrin<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, December 01, 2008
4:23 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Inbound
calls from Asterisk to Asterisk with SIP"Forbidden" from
'"asterisk"</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><span class=value><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>Please help.</span></font></span><br>
<br>
<span class=value>Asterisk 1: Sip.conf</span><br>
<span class=value>[VoipDirect777821]</span><br>
<span class=value>type=friend</span><br>
<span class=value>host=dfvvd.dyndns.org</span><br>
<span class=value>username=VoipDirect777821</span><br>
<span class=value>secret=xxxxxxxxxxxx</span><br>
<span class=value>accountcode=5260477782</span><br>
<span class=value>amaflags=billing</span><br>
<span class=value>context=Incoming</span><br>
<span class=value>disallow=all</span><br>
<span class=value>allow=g729</span><br>
<span class=value>;allow=alaw</span><br>
<span class=value>;allow=ulaw</span><br>
<span class=value>trunk=no</span><br>
<span class=value>qualify=yes</span><br>
<span class=value>qualifysmoothing=yes</span><br>
<span class=value>nat=no</span><br>
<span class=value>canreinvite=yes</span><br>
<span class=value>dtmfmode=rfc2833</span><br>
<span class=value>;directrtpsetup=no</span><br>
<span class=value>t38pt_udptl = yes</span><br>
<br>
<span class=value>Asterisk 2 sip.conf</span><br>
<span class=value> GNU nano 1.3.12
File: sip_custom.conf</span><br>
<br>
<span class=value>[VoipDirect777821]</span><br>
<span class=value>type=friend</span><br>
<span class=value>host=141.122.139</span><br>
<span class=value>username=VoipDirect777821</span><br>
<span class=value>secret=wsPiOov8830</span><br>
<span class=value>accountcode=5260477782</span><br>
<span class=value>amaflags=billing</span><br>
<span class=value>context=Incomming</span><br>
<span class=value>disallow=all</span><br>
<span class=value>allow=g729</span><br>
<span class=value>;allow=alaw</span><br>
<span class=value>;allow=ulaw</span><br>
<span class=value>trunk=no</span><br>
<span class=value>qualify=yes</span><br>
<span class=value>qualifysmoothing=yes</span><br>
<span class=value>nat=no</span><br>
<span class=value>canreinvite=yes</span><br>
<span class=value>dtmfmode=rfc2833</span><br>
<span class=value>;directrtpsetup=no</span><br>
<span class=value>t38pt_udptl = yes</span><br>
<br>
<span class=value>sip show peers shows both as registered.</span><br>
<br>
<span class=value>this is the error when try and place a call from Asterisk 1
to Asterisk 2:</span><br>
<br>
<span class=value>- Executing [582@a1:1] Dial("Console/dsp",
"SIP/VoipDirect777821|60|"<wbr>) in new stack</span><br>
<span class=value> -- Called VoipDirect777821</span><br>
<span class=value>[Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334
handle_response_invite: Received response: "Forbidden" from
'"asterisk" <sip:asterisk@141.122.139.<wbr>16>;tag=as<wbr>070b02e2'</span><br>
<span class=value> -- SIP/VoipDirect777821-0876c<wbr>360 is
circuit-busy</span><br>
<span class=value> == Everyone is busy/congested at this time (1:0/1/0)</span><br>
<span class=value> -- Executing [582@a1:2]
Hangup("Console/dsp", "") in new stack</span><br>
<span class=value> == Spawn extension (a1, 582, 2) exited non-zero on
'Console/dsp'</span><br>
<span class=value> << Hangup on console >></span><br>
<br>
<span class=value>I get the same error even if I include this on Asterisk 1:</span><br>
<span class=value>register => VoipDirect777821:xxxxx@dfv<wbr>vd.dyndns.<wbr>org</span><br>
<br>
<span class=value>Please help....</span> <o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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