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

[James A. Shigley]

What do you see when you run asterisk –r and dial 210 or 211 from one of the phones

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

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From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk

 

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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