Before you read… please understand that i’m new and i have no place to refer to regarding this topic… thanks in advance.
Hello to you all asterisk users.
Being new to this topic, as usual i started reading. So i read. After some time, it occured to me to try one of the configurations that the book was explaining(asterisk - the future of telephony), but the configuration didn’t work. I tried looking on the net for other examples, but they were all just like the one in the book, so it must be me not doing the right things. The aterisk PBX does not open the 5060 port for a sip.conf configuration. Here’s the code:
sip.conf===================start
[general]
port=5060
bindaddr=0.0.0.0
context=default
srvlookup=yes
[john]
type=friend
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
sip.conf====================end
extensions.conf===============start
[general]
[internal]
exten => 611,1,answer
exten => 611,2,echo
[local]
exten => 611,1,answer
exten => 611,2,echo
extensions.conf===============end
This configuration is for testing that the PBX is working.
The PBX’s ip address is 192.168.0.5 with no firewall
The client’s ip address is 192.168.0.4, using x-lite, no firewall here also.
As you can see the network is not NAT-ed.
I tried: dial 611, and worked only from the CLI, but xlite didn’t work.
After trying several times, there was no succes.
I nmap-ed the server and discovered that no port was open by the PBX, 5060 respectively (TCP or UDP): nmap 192.168.0.5 - no sucess, no port open
netstat | grep asterisk - again no port open, be it TCP or UDP.
The calls work only from the PBX’s CLI.
Xlite’s configuration was the following:
Display name:
Usename:john
Authorization user:john
password:welcome
domain/realm:192.168.0.5
SIP proxy:192.168.0.5:5060
the rest are unchanged and remain at their default value.
I downloaded a linux version of xlite, and configured it same as above, so now i had both the server and the client on the same ip: 192.168.0.5, but still no luck. I’m stuck
I compiled the asterisk PBX with the default values, and modified the extensions.conf and sip.conf files to what you see above.
What am i missing here?
Thanks a bunch.