May I suggest divide and conquer?
I haven't followed every detail, but it seems that your phones are not registering. Put them on the same net as the sip server and get them to register. Then get it to where you can make a call from one to the other. Then back off through your router or what ever, with what ever filtering you have in place. In other words get it working "up close" then move out. until you break it, and find that problem. 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. I hope this helps. If I have missed the mark, explain more as to the point it is failing. Cary Fitch _____ From: [email protected] [mailto:[email protected]] On Behalf Of jonas kellens Sent: Tuesday, April 14, 2009 12:58 PM To: [email protected] Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk 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.
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