i am not sure what you are talking about. i have extensions and my sip trunk
config in that file. see below
[200]
deny=0.0.0.0/0.0.0.0
type=friend
secret=200
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=...@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/200
context=from-internal
canreinvite=yes
callgroup=
callerid=device <200>
accountcode=
call-limit=50
[BW-SIP-A]
disallow=all
canreinvite=yes
dtmfmode=rfc2833
host=x.x.x.x
outboundproxy=x.x.x.x
progressinbound=yes
qualify=300
type=peer
allow=ulaw
[BW-SIP-B]
disallow=all
canreinvite=yes
dtmfmode=rfc2833
host=x.x.x.x
outboundproxy=x.x.x.x
progressinbound=yes
qualify=300
type=peer
allow=ulaw
[from-bandwidth-A]
disallow=all
type=peer
reinvite=yes
port=5060
insecure=invite,port
host=x.x.x.x
fromdomain=x.x.x.x
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
qualify=300
[from-bandwidth-B]
disallow=all
type=peer
reinvite=yes
port=5060
insecure=invite,port
host=x.x.x.x
fromdomain=x.x.x.x
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
qualify=300
Date: Fri, 14 Aug 2009 12:09:15 -0500
From: crt.ro...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no ring tone
Hello
One question
In sip.con or sip_additionals.conf, in freepbx, the context of your client do
you put
nat = yes
externip = XXXX
You put your public ip.
Are you sure that?
Regards
On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose <sixfourimp...@hotmail.com> wrote:
i changed it and still didn't ring. however it did ring on one call to a cell
phone but it hasn't done it again.
Date: Fri, 14 Aug 2009 09:39:33 -0500
From: crt.ro...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no ring tone
Hello,
I never use externhost
y use \
externip=public ip
And work fine
Regards
On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose <sixfourimp...@hotmail.com> wrote:
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external
hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your external domain name here)
nat=yes
qualify=yes
canreinvite=no
Add extra codecs to /etc/asterisk/sip_custom.conf
allow=gsm allow=h261
allow=h263
allow=h263p
videosupport=yes
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