[asterisk-users] no ring tone

[Ott Rose]

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=200 at 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.rojas at gmail.com
To: asterisk-users at 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 <sixfourimpala at 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.rojas at gmail.com

To: asterisk-users at 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 <sixfourimpala at 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
Windows Live™: Keep your life in sync. Check it out.


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