[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk

Casey Boone cboone at shawneelink.net
Tue Apr 14 10:36:43 CDT 2009


just for a test, run "service iptables stop" as root on the asterisk 
server and then reboot your phones.  after that, try again and see if 
the phones are making communications with asterisk.

you can turn the firewall back on with "service iptables start"

jonas kellens wrote:
> Hi there,
> 
> this is the first time that I'm building an Asterisk-server.
> 
> I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
> Zaptel is for later, when configuring the POTS-line. Now first internal 
> communication with SIP.
> 
> Thought it would go easier...
> 
> I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
> 
> These are my settings :
> 
> sip.conf :
> /[root at asterisk asterisk]# cat sip.conf/
> /[general]/
> /bindport=5060/
> /bindaddr = 0.0.0.0/
> 
> /[BT201]/
> /type=friend/
> /context=intern/
> /host=192.168.4.210/
> /secret=testpaswoord/
> 
> /[GXP1200]/
> /type=friend/
> /context=intern/
> /host=192.168.4.211/
> /secret=testpaswoord/
> extensions.conf :
> /[root at asterisk asterisk]# cat extensions.conf/
> /[intern]/
> /exten => 210,1,Dial(SIP/BT201)/
> /exten => 211,1,Dial(SIP/GXP1200)/
> Asterisk CLI shows me :
> /asterisk*CLI> sip reload/
> /Reloading SIP/
> /  == Parsing '/etc/asterisk/sip.conf': Found/
> /  == Parsing '/etc/asterisk/users.conf': Found/
> /  == Parsing '/etc/asterisk/sip_notify.conf': Found/
> /asterisk*CLI> sip show peers/
> /Name/username              Host            Dyn Nat ACL Port     Status  
>              /
> /GXP1200                    192.168.4.211               5060    
>  Unmonitored           /
> /BT201                      192.168.4.210               5060    
>  Unmonitored           /
> /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 
> offline]/
> 
> /asterisk*CLI> dialplan show intern/
> /[ Context 'intern' created by 'pbx_config' ]/
> /  '210' =>          1. Dial(SIP/BT201)                            
> [pbx_config]/
> /  '211' =>          1. Dial(SIP/GXP1200)                          
> [pbx_config]/
> 
> I pick up the phone of the BT201 and dial 211... nothing happens.
> I pick up the phone of the GXP1200 and dial 210... nothing happens.
> 
> I would love to have your feedback on this. Where could this problem be 
> situated ?
> 
> I notice (on the Asterisk CLI) that my SIP-phones do not register. They 
> have a fixed IP and there account information is set via the web interface.
> 
> Greetingz,
> Jonas.
> 
> 
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