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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'>May I suggest divide and conquer? <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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I haven’t followed every detail, but
it seems that your phones are not registering.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Put them on the same net as the sip server
and get them to register.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Then get it to where you can make a call
from one to the other.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Then back off through your router or what
ever, with what ever filtering you have in place.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>In other words get it working “up
close” then move out… until you break it, and find that problem.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>With at least one phone working, you can
be sure the system is “good” at any instant and then make the other
“distant” phone work too.<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>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I hope this helps. If I have missed
the mark, explain more as to the point it is failing.<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>
<p class=MsoNormal><st1:City w:st="on"><st1:place w:st="on"><font size=2
color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>Cary</span></font></st1:place></st1:City><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'> Fitch<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>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<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>jonas kellens<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, April 14, 2009
12:58 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> Re: [asterisk-users]
Asterisk-beginner : cannot make phonecalls using 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><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I will summarize everything again and try to answer all the
questions asked while I was away.<br>
<br>
First I stop Asterisk :<br>
<br>
</span></font><i><font size=2 color=blue><span style='font-size:10.0pt;
color:blue;font-style:italic'>[root@asterisk asterisk]# /usr/sbin/asterisk -r</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Created by Mark Spencer <<a href="mailto:markster@digium.com">markster@digium.com</a>></span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>This is free software, with components licensed under the GNU General
Public</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>License version 2 and other licenses; you are welcome to redistribute
it under</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>certain conditions. Type 'core show license' for details.</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>=========================================================================</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3189)</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Verbosity is at least 3</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>asterisk*CLI> stop now</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>asterisk*CLI> </span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>Disconnected from Asterisk server</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>[root@asterisk asterisk]# ps aux | grep asterisk</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>avahi 3320 0.0 0.0
2588 1344 ? Ss
18:49 0:00 avahi-daemon: running [asterisk.local]</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>root 3563 0.0 0.0
3912 676 pts/0 S+ 19:11
0:00 grep asterisk</span></font></i><br>
<br>
Then I edit the files sip.conf and extensions.conf<br>
<br>
<u>SIP.CONF</u><br>
<br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>[root@asterisk asterisk]# cat sip.conf</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>[general]</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>context=default</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>port=5060</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>bindaddr=192.168.4.248</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>srvlookup=yes</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>disallow=all</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>allow=ulaw</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>allow=gsm</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>allow=g711</span></font></i><br>
<br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>[BT201]</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>type=friend</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>context=intern</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>host=dynamic</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>username=BT201</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>secret=testpaswoord</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>;canreinvite=yes</span></font></i><br>
<br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>[GXP1200]</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>type=friend</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>context=intern</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>host=dynamic</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>username=GXP1200</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>secret=testpaswoord</span></font></i><br>
<i><font size=2 color=blue><span style='font-size:10.0pt;color:blue;font-style:
italic'>;canreinvite=yes</span></font></i><br>
<br>
<u>EXTENSIONS.CONF</u><br>
<br>
<i><font color=blue><span style='color:blue;font-style:italic'>[root@asterisk
asterisk]# cat extensions.conf</span></font></i><br>
<i><font color=blue><span style='color:blue;font-style:italic'>[globals]</span></font></i><br>
<br>
<i><font color=blue><span style='color:blue;font-style:italic'>[default]</span></font></i><br>
<br>
<i><font color=blue><span style='color:blue;font-style:italic'>[intern]</span></font></i><br>
<i><font color=blue><span style='color:blue;font-style:italic'>exten =>
210,1,Dial(SIP/BT201,30)</span></font></i><br>
<i><font color=blue><span style='color:blue;font-style:italic'>exten =>
211,1,Dial(SIP/GXP1200,30)</span></font></i><br>
<br>
<i><font color=blue><span style='color:blue;font-style:italic'>exten =>
251,1,Answer()</span></font></i><br>
<i><font color=blue><span style='color:blue;font-style:italic'>exten =>
251,n,Echo()</span></font></i><br>
<i><font color=blue><span style='color:blue;font-style:italic'>exten =>
251,n,Hangup()</span></font></i><br>
<br>
Then I configure my SIP-phone grandstream BT201 :<br>
<br>
<font color=blue><span style='color:blue'>1) I press menu > dhcp [on]</span></font><br>
<font color=blue><span style='color:blue'>2) I press menu > IP-address >
192.168.4.144</span></font><br>
<font color=blue><span style='color:blue'>3) I go to the webinterface via the
above IP-address</span></font><br>
<b><font color=blue><span style='color:blue;font-weight:bold'>My settings :</span></font></b><br>
<font color=blue><span style='color:blue'>> tab <u>account</u></span></font><br>
<font color=blue><span style='color:blue'>account name : BT201</span></font><br>
<font color=blue><span style='color:blue'>SIP server : 192.168.4.248</span></font><br>
<font color=blue><span style='color:blue'>Outbound proxy : 192.168.4.248</span></font><br>
<font color=blue><span style='color:blue'>SIP user ID : BT201</span></font><br>
<font color=blue><span style='color:blue'>Authenticate ID : BT201</span></font><br>
<font color=blue><span style='color:blue'>Authenticate Password : testpaswoord</span></font><br>
<font color=blue><span style='color:blue'>Name : BT201</span></font><br>
<font color=blue><span style='color:blue'>Use DNS SRV : no</span></font><br>
<font color=blue><span style='color:blue'>User ID is phone number : no</span></font><br>
<font color=blue><span style='color:blue'>SIP registration : yes</span></font><br>
<font color=blue><span style='color:blue'>Unregister on reboot : no</span></font><br>
<font color=blue><span style='color:blue'>Register expiration : 60</span></font><br>
<font color=blue><span style='color:blue'>local SIP port : 5060</span></font><br>
<font color=blue><span style='color:blue'>SIP transport : UDP</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>Use RFC3581
Symmetric Routing</span></font></i><font color=blue><span style='color:blue'> :
no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>NAT Traversal
(STUN)</span></font></i><font color=blue><span style='color:blue'> : no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>SUBSCRIBE for
MWI</span></font></i><font color=blue><span style='color:blue'> : no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>Proxy-Require</span></font></i><font
color=blue><span style='color:blue'> : (nothing)</span></font><br>
<br>
<font color=blue><span style='color:blue'>> Update > Reboot</span></font><br>
<br>
Then I configure my SIP-phone grandstream GX1200 :<br>
<br>
<br>
<font color=blue><span style='color:blue'>1) I press menu > status</span></font><br>
<font color=blue><span style='color:blue'>2) IP-address : 192.168.4.180</span></font><br>
<font color=blue><span style='color:blue'>3) I go to the webinterface via the
above IP-address</span></font><br>
<b><font color=blue><span style='color:blue;font-weight:bold'>My settings :</span></font></b><br>
<font color=blue><span style='color:blue'>> tab <u>account</u></span></font><br>
<font color=blue><span style='color:blue'>account 1 active : yes</span></font><br>
<font color=blue><span style='color:blue'>account name : GX1200</span></font><br>
<font color=blue><span style='color:blue'>SIP server : 192.168.4.248</span></font><br>
<font color=blue><span style='color:blue'>Outbound proxy : 192.168.4.248</span></font><br>
<font color=blue><span style='color:blue'>SIP user ID : GX1200</span></font><br>
<font color=blue><span style='color:blue'>Authenticate ID : GX1200</span></font><br>
<font color=blue><span style='color:blue'>Authenticate Password : testpaswoord</span></font><br>
<font color=blue><span style='color:blue'>Name : GX1200</span></font><br>
<font color=blue><span style='color:blue'>Use DNS SRV : no</span></font><br>
<font color=blue><span style='color:blue'>User ID is phone number : no</span></font><br>
<font color=blue><span style='color:blue'>SIP registration : yes</span></font><br>
<font color=blue><span style='color:blue'>Unregister on reboot : no</span></font><br>
<font color=blue><span style='color:blue'>Register expiration : 60</span></font><br>
<font color=blue><span style='color:blue'>local SIP port : 5060</span></font><br>
<font color=blue><span style='color:blue'>SIP transport : UDP</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>Use RFC3581
Symmetric Routing</span></font></i><font color=blue><span style='color:blue'> :
no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>NAT Traversal
(STUN)</span></font></i><font color=blue><span style='color:blue'> : no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>SUBSCRIBE for
MWI</span></font></i><font color=blue><span style='color:blue'> : no</span></font><br>
<i><font color=blue><span style='color:blue;font-style:italic'>Proxy-Require</span></font></i><font
color=blue><span style='color:blue'> : (nothing)</span></font><br>
<br>
Then I unplug the power of the Grandstream IP-telephones.<br>
<br>
I restart Asterisk on my server :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[root@asterisk
asterisk]# /sbin/service asterisk start</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Starting
asterisk:
[ OK ]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[root@asterisk
asterisk]# /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Asterisk
1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Created by
Mark Spencer <<a href="mailto:markster@digium.com">markster@digium.com</a>></span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Asterisk
comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>This is free
software, with components licensed under the GNU General Public</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>License
version 2 and other licenses; you are welcome to redistribute it under</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>certain
conditions. Type 'core show license' for details.</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>=========================================================================</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'> ==
Parsing '/etc/asterisk/asterisk.conf': Found</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'> ==
Parsing '/etc/asterisk/extconfig.conf': Found</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Connected to
Asterisk 1.4.24 currently running on asterisk (pid = 3683)</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Verbosity was
3 and is now 34</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
</span></font><br>
<br>
I wait a while but no output on the CLI...<br>
<br>
Then I give some commands :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
sip show peers</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port Status
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>GXP1200/GXP1200
(Unspecified)
D
0 Unmonitored
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>BT201/BT201
(Unspecified)
D
0 Unmonitored
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>2 sip peers
[Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</span></font><br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
sip debug</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>SIP Debugging
enabled</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>The 'sip
debug' command is deprecated and will be removed in a future release. Please
use 'sip set debug' instead.</span></font><br>
<br>
Then I power back on my Grandstream IP-telephones.<br>
<br>
Nothing happens on the CLI...<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
sip show peers</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port
Status
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>GXP1200/GXP1200
(Unspecified)
D
0 Unmonitored
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>BT201/BT201
(Unspecified) D
0
Unmonitored </span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>2 sip peers
[Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</span></font><br>
<br>
My iptables settings :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[root@asterisk
sysconfig]# cat iptables</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'># Firewall
configuration written by system-config-securitylevel</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'># Manual
customization of this file is not recommended.</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>*filter</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>:INPUT ACCEPT
[0:0]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>:FORWARD
ACCEPT [0:0]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>:OUTPUT
ACCEPT [0:0]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>:RH-Firewall-1-INPUT
- [0:0]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A INPUT -j
RH-Firewall-1-INPUT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A FORWARD -j
RH-Firewall-1-INPUT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -i lo -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A RH-Firewall-1-INPUT
-p icmp --icmp-type any -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p 50 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p 51 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p udp --dport 5353 -d 224.0.0.251 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p tcp -m tcp --dport 631 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>-A
RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>COMMIT</span></font><br>
<br>
I added the line "<font size=2 color=blue><span style='font-size:10.0pt;
color:blue'>-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT</span></font>"
to the file...<br>
<br>
Netstat :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[root@asterisk
sysconfig]# netstat -a -n -p | grep 5060</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>udp
0 0
192.168.4.248:5060
0.0.0.0:*
3683/asterisk </span></font><br>
<br>
TCPdump :<br>
<br>
I put the power off and back on of the IP-phones, otherwise nothing happens :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[root@asterisk
sysconfig]# /usr/sbin/tcpdump port 5060</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>tcpdump: verbose
output suppressed, use -v or -vv for full protocol decode</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>listening on
eth1, link-type EN10MB (Ethernet), capture size 96 bytes</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:33.106887
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:34.106254
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:36.106065
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:37.343330
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:38.342736
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:40.105688
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:47:40.342297
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:14.071499
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:14.819554
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:15.068907
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:15.816712
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:17.068718
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:17.816524
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:21.068341
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:21.816147
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:25.067975
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:25.815769
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:49.066450
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:49.814257
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:50.065855
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:50.813411
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:52.065667
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:52.813473
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:56.065290
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:48:56.813095
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:49:00.064913
IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>19:49:00.812718
IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</span></font><br>
<br>
Meanwhile the Grandstream IP-phones have powered up...<br>
So on port 5060, there are packets that arrive...<br>
<br>
Does my Asterisk really listen on 5060 ??<br>
<br>
Are my iptables configured the right way ??<br>
<br>
A last test + output on the CLI :<br>
<br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Connected to
Asterisk 1.4.24 currently running on asterisk (pid = 3683)</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Verbosity is
at least 34</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
originate SIP/BT201 application playback demo-instruct</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Really
destroying SIP dialog '0e4fed4c60b54b44169dad7a0f84ca98@192.168.4.248' Method:
INVITE</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>[Apr 14
19:54:04] NOTICE[3763]: channel.c:3033 __ast_request_and_dial: Unable to
request channel SIP/BT201</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
sip show peers</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port
Status
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>GXP1200/GXP1200
(Unspecified)
D
0 Unmonitored
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>BT201/BT201
(Unspecified)
D
0 Unmonitored
</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>2 sip peers
[Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</span></font><br>
<font size=2 color=blue><span style='font-size:10.0pt;color:blue'>asterisk*CLI>
</span></font><br>
<br>
<br>
Thanks to everyone who is trying to help me !! Sincerely !<br>
<br>
Jonas. <o:p></o:p></p>
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