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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>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk asterisk]# /usr/sbin/asterisk -r</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Created by Mark Spencer <<A HREF="mailto:markster@digium.com">markster@digium.com</A>></FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">This is free software, with components licensed under the GNU General Public</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">License version 2 and other licenses; you are welcome to redistribute it under</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">certain conditions. Type 'core show license' for details.</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">=========================================================================</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3189)</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Verbosity is at least 3</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> stop now</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> </FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">Disconnected from Asterisk server</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk asterisk]# ps aux | grep asterisk</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">avahi 3320 0.0 0.0 2588 1344 ? Ss 18:49 0:00 avahi-daemon: running [asterisk.local]</FONT></FONT></I><BR>
<I><FONT COLOR="#0000ff"><FONT SIZE="2">root 3563 0.0 0.0 3912 676 pts/0 S+ 19:11 0:00 grep asterisk</FONT></FONT></I><BR>
<BR>
Then I edit the files sip.conf and extensions.conf<BR>
<BR>
<U>SIP.CONF</U><BR>
<BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">[root@asterisk asterisk]# cat sip.conf</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">[general]</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">context=default</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">port=5060</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">bindaddr=192.168.4.248</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">srvlookup=yes</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">disallow=all</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">allow=ulaw</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">allow=gsm</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">allow=g711</FONT></I></FONT><BR>
<BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">[BT201]</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">type=friend</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">context=intern</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">host=dynamic</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">username=BT201</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">secret=testpaswoord</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">;canreinvite=yes</FONT></I></FONT><BR>
<BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">[GXP1200]</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">type=friend</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">context=intern</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">host=dynamic</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">username=GXP1200</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">secret=testpaswoord</FONT></I></FONT><BR>
<FONT SIZE="2"><I><FONT COLOR="#0000ff">;canreinvite=yes</FONT></I></FONT><BR>
<BR>
<U>EXTENSIONS.CONF</U><BR>
<BR>
<I><FONT COLOR="#0000ff">[root@asterisk asterisk]# cat extensions.conf</FONT></I><BR>
<I><FONT COLOR="#0000ff">[globals]</FONT></I><BR>
<BR>
<I><FONT COLOR="#0000ff">[default]</FONT></I><BR>
<BR>
<I><FONT COLOR="#0000ff">[intern]</FONT></I><BR>
<I><FONT COLOR="#0000ff">exten => 210,1,Dial(SIP/BT201,30)</FONT></I><BR>
<I><FONT COLOR="#0000ff">exten => 211,1,Dial(SIP/GXP1200,30)</FONT></I><BR>
<BR>
<I><FONT COLOR="#0000ff">exten => 251,1,Answer()</FONT></I><BR>
<I><FONT COLOR="#0000ff">exten => 251,n,Echo()</FONT></I><BR>
<I><FONT COLOR="#0000ff">exten => 251,n,Hangup()</FONT></I><BR>
<BR>
Then I configure my SIP-phone grandstream BT201 :<BR>
<BR>
<FONT COLOR="#0000ff">1) I press menu > dhcp [on]</FONT><BR>
<FONT COLOR="#0000ff">2) I press menu > IP-address > 192.168.4.144</FONT><BR>
<FONT COLOR="#0000ff">3) I go to the webinterface via the above IP-address</FONT><BR>
<B><FONT COLOR="#0000ff">My settings :</FONT></B><BR>
<FONT COLOR="#0000ff">> tab </FONT><FONT COLOR="#0000ff"><U>account</U></FONT><BR>
<FONT COLOR="#0000ff">account name : BT201</FONT><BR>
<FONT COLOR="#0000ff">SIP server : 192.168.4.248</FONT><BR>
<FONT COLOR="#0000ff">Outbound proxy : 192.168.4.248</FONT><BR>
<FONT COLOR="#0000ff">SIP user ID : BT201</FONT><BR>
<FONT COLOR="#0000ff">Authenticate ID : BT201</FONT><BR>
<FONT COLOR="#0000ff">Authenticate Password : testpaswoord</FONT><BR>
<FONT COLOR="#0000ff">Name : BT201</FONT><BR>
<FONT COLOR="#0000ff">Use DNS SRV : no</FONT><BR>
<FONT COLOR="#0000ff">User ID is phone number : no</FONT><BR>
<FONT COLOR="#0000ff">SIP registration : yes</FONT><BR>
<FONT COLOR="#0000ff">Unregister on reboot : no</FONT><BR>
<FONT COLOR="#0000ff">Register expiration : 60</FONT><BR>
<FONT COLOR="#0000ff">local SIP port : 5060</FONT><BR>
<FONT COLOR="#0000ff">SIP transport : UDP</FONT><BR>
<I><FONT COLOR="#0000ff">Use RFC3581 Symmetric Routing</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">NAT Traversal (STUN)</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">SUBSCRIBE for MWI</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">Proxy-Require</FONT></I><FONT COLOR="#0000ff"> : (nothing)</FONT><BR>
<BR>
<FONT COLOR="#0000ff">> Update > Reboot</FONT><BR>
<BR>
Then I configure my SIP-phone grandstream GX1200 :<BR>
<BR>
<BR>
<FONT COLOR="#0000ff">1) I press menu > status</FONT><BR>
<FONT COLOR="#0000ff">2) IP-address : 192.168.4.180</FONT><BR>
<FONT COLOR="#0000ff">3) I go to the webinterface via the above IP-address</FONT><BR>
<B><FONT COLOR="#0000ff">My settings :</FONT></B><BR>
<FONT COLOR="#0000ff">> tab </FONT><FONT COLOR="#0000ff"><U>account</U></FONT><BR>
<FONT COLOR="#0000ff">account 1 active : yes</FONT><BR>
<FONT COLOR="#0000ff">account name : GX1200</FONT><BR>
<FONT COLOR="#0000ff">SIP server : 192.168.4.248</FONT><BR>
<FONT COLOR="#0000ff">Outbound proxy : 192.168.4.248</FONT><BR>
<FONT COLOR="#0000ff">SIP user ID : GX1200</FONT><BR>
<FONT COLOR="#0000ff">Authenticate ID : GX1200</FONT><BR>
<FONT COLOR="#0000ff">Authenticate Password : testpaswoord</FONT><BR>
<FONT COLOR="#0000ff">Name : GX1200</FONT><BR>
<FONT COLOR="#0000ff">Use DNS SRV : no</FONT><BR>
<FONT COLOR="#0000ff">User ID is phone number : no</FONT><BR>
<FONT COLOR="#0000ff">SIP registration : yes</FONT><BR>
<FONT COLOR="#0000ff">Unregister on reboot : no</FONT><BR>
<FONT COLOR="#0000ff">Register expiration : 60</FONT><BR>
<FONT COLOR="#0000ff">local SIP port : 5060</FONT><BR>
<FONT COLOR="#0000ff">SIP transport : UDP</FONT><BR>
<I><FONT COLOR="#0000ff">Use RFC3581 Symmetric Routing</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">NAT Traversal (STUN)</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">SUBSCRIBE for MWI</FONT></I><FONT COLOR="#0000ff"> : no</FONT><BR>
<I><FONT COLOR="#0000ff">Proxy-Require</FONT></I><FONT COLOR="#0000ff"> : (nothing)</FONT><BR>
<BR>
Then I unplug the power of the Grandstream IP-telephones.<BR>
<BR>
I restart Asterisk on my server :<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk asterisk]# /sbin/service asterisk start</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Starting asterisk: [ OK ]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk asterisk]# /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Created by Mark Spencer <<A HREF="mailto:markster@digium.com">markster@digium.com</A>></FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">This is free software, with components licensed under the GNU General Public</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">License version 2 and other licenses; you are welcome to redistribute it under</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">certain conditions. Type 'core show license' for details.</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">=========================================================================</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> == Parsing '/etc/asterisk/asterisk.conf': Found</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"> == Parsing '/etc/asterisk/extconfig.conf': Found</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3683)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Verbosity was 3 and is now 34</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> </FONT></FONT><BR>
<BR>
I wait a while but no output on the CLI...<BR>
<BR>
Then I give some commands :<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> sip show peers</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Name/username Host Dyn Nat ACL Port Status </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">GXP1200/GXP1200 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">BT201/BT201 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</FONT></FONT><BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> sip debug</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">SIP Debugging enabled</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.</FONT></FONT><BR>
<BR>
Then I power back on my Grandstream IP-telephones.<BR>
<BR>
Nothing happens on the CLI...<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">asterisk*CLI> sip show peers</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">Name/username Host Dyn Nat ACL Port Status </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">GXP1200/GXP1200 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">BT201/BT201 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</FONT></FONT><BR>
<BR>
My iptables settings :<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk sysconfig]# cat iptables</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"># Firewall configuration written by system-config-securitylevel</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2"># Manual customization of this file is not recommended.</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">*filter</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">:INPUT ACCEPT [0:0]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">:FORWARD ACCEPT [0:0]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">:OUTPUT ACCEPT [0:0]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">:RH-Firewall-1-INPUT - [0:0]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A INPUT -j RH-Firewall-1-INPUT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A FORWARD -j RH-Firewall-1-INPUT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -i lo -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p icmp --icmp-type any -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p 50 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p 51 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p udp --dport 5353 -d 224.0.0.251 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p tcp -m tcp --dport 631 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">COMMIT</FONT></FONT><BR>
<BR>
I added the line "<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT</FONT></FONT>" to the file...<BR>
<BR>
Netstat :<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk sysconfig]# netstat -a -n -p | grep 5060</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">udp 0 0 192.168.4.248:5060 0.0.0.0:* 3683/asterisk </FONT></FONT><BR>
<BR>
TCPdump :<BR>
<BR>
I put the power off and back on of the IP-phones, otherwise nothing happens :<BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[root@asterisk sysconfig]# /usr/sbin/tcpdump port 5060</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">tcpdump: verbose output suppressed, use -v or -vv for full protocol decode</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:33.106887 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:34.106254 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:36.106065 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:37.343330 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:38.342736 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:40.105688 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:47:40.342297 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:14.071499 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:14.819554 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:15.068907 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:15.816712 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:17.068718 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:17.816524 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:21.068341 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:21.816147 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:25.067975 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:25.815769 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:49.066450 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:49.814257 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:50.065855 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:50.813411 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:52.065667 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:52.813473 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:56.065290 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:48:56.813095 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:49:00.064913 IP 192.168.4.144.sip > 192.168.4.248.sip: SIP, length: 505</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">19:49:00.812718 IP 192.168.4.180.sip > 192.168.4.248.sip: SIP, length: 523</FONT></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"><FONT COLOR="#0000ff">Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3683)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">Verbosity is at least 34</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">asterisk*CLI> originate SIP/BT201 application playback demo-instruct</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">Really destroying SIP dialog '0e4fed4c60b54b44169dad7a0f84ca98@192.168.4.248' Method: INVITE</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Apr 14 19:54:04] NOTICE[3763]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/BT201</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">asterisk*CLI> sip show peers</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">Name/username Host Dyn Nat ACL Port Status </FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">GXP1200/GXP1200 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">BT201/BT201 (Unspecified) D 0 Unmonitored </FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">asterisk*CLI> </FONT></FONT><BR>
<BR>
<BR>
Thanks to everyone who is trying to help me !! Sincerely !<BR>
<BR>
Jonas.
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