Hi all,
I have started with AAH because I don’t know linux well yet and the packaged, fire 'n forget installation has made it easier to get started for me.
I also ran up to Barnes & Noble and grabbed a copy of the O’Reily book yesterday. I’ve read through it a few times. (Asterisk: The future of telephony. Great book, btw. Explains the concepts of asterisk and it’s operation very well.). I would like to start following some of the examples in the book and playing with coding dial plans by hand.
The problem that I am running into is successfully commenting out (or altogether excluding) the AMP/FeePBX include’s, etc. What I ended up with after backing up the originals was the following sip.conf:
[code][general]
externip=222.222.222.222
localnet=192.168.0.0/255.255.255.0
nat=yes
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context=test
callerid = Unknown
;**************************************
;Registration statement
;**************************************
register=MyUserName:MyPassword@sip.axvoice.com
; same record from sip_additional.conf file.
; just used here so to test without having to have
; the includes below, which have all AMP/FreePBX data.
[1000]
username=1000
type=friend
secret=abc123
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=no
mailbox=1000@device
host=192.168.1.105
dtmfmode=rfc2833
context=test
canreinvite=no
callerid=Harry Keogh <1000>
[axVoice]
username=MyUserName
type=friend
secret=MyPassword
insecure=very
host=216.143.130.36
fromuser=MyUserName
fromdomain=216.143.130.36
dtmfmode=rfc2833
disallow=all
defaultip=216.143.130.36
context=from-pstn
canreinvite=no
authname=MyUserName
allow=true
;******************
;comment out includes
;******************
;#include sip_nat.conf
;#include sip_custom.conf
;#include sip_additional.conf[/code]
I also have the following extensions.conf file after backing up the original:
[test]
exten s,1,Answer()
exten s,2,Playback(hello-world)
exten s,3,Hangup()
The first problem that I have is that asterisk will no longer register with my provider (check in Maintenance > Asterisk Info). I thought that sip_additional.conf was used only by AMP so I could just by-pass the include’s and put the registration statement in the same sip.conf file.
The second problem, I don’t know what it is yet. I’m just preping everyone for my next post
Is there anything glaringly obvious that I’m missing in the .conf files shown here?
Thanks,
Lee