[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
jonas kellens
jonas.kellens at telenet.be
Tue Apr 14 12:57:31 CDT 2009
I will summarize everything again and try to answer all the questions
asked while I was away.
First I stop Asterisk :
[root at asterisk asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
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
[root at 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
[root at 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
[root at 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 :
[root at asterisk asterisk]# /sbin/service asterisk start
Starting asterisk: [ OK ]
[root at asterisk asterisk]# /usr/sbin/asterisk
-vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
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 :
[root at 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 :
[root at 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 :
[root at 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
'0e4fed4c60b54b44169dad7a0f84ca98 at 192.168.4.248' 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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