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