[Asterisk-bsd] Re: One Way Audio
Jay Adelson
jay at adelson.org
Sat Oct 16 08:25:05 CDT 2004
I'll check the codecs again. Meanwhile, here is the cleaned sip.conf.
-Jay
[general]
port = 5060
bindaddr= 0.0.0.0
externip = 1.2.3.4
localnet = 192.168.168.0/255.255.255.0 ;local network and mask
context = from-sip ; Default for incoming calls
callerid=No CallID
;Register for sipphone.com:
register=17471234567:my-pass at proxy01.sipphone.com/17471234567
[proxy01.sipphone.com]
type=friend
secret=my-pass
username=17471234567
host=proxy01.sipphone.com
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=g729
fromdomain=sipphone.com
callerid="Me <1747123456789>"
insecure=very
qualify=no
reinvite=no
canreinvite=no
nat=yes
[1000]
type=friend
username=1000
password=polycompass
host=dynamic
defaultip=192.168.168.50
mailbox=101
context=intern
reinvite=yes
canreinvite=yes
dtmfmode=inband
nat=0
[1001]
type=friend
authname=1001
secret=polycompass
host=dynamic
defaultip=192.168.168.50
mailbox=102
context=intern
canreinvite=yes
dtmfmode=inband
nat=0
[1002]
type=friend
username=1002
password=sjphonepass
host=dynamic
mailbox=103
context=intern
canreinvite=no
nat=0
On Sat, Oct 16, 2004 at 10:01:25AM +0200, Tom Ivar Helbekkmo wrote:
> Jay Adelson <jay at adelson.org> writes:
>
> > If the remote side calls my phone, particularly with reinvite on, I
> > can hear the remote side fine and the call works normally.
>
> Well, since the call runs through an Asterisk that has one foot on the
> outside, with real addresses, and one on the inside, with the phones
> on, and NAT in between, it's important that canreinvite=no. Otherwise,
> the phone end points will try to speak RTP directly, and while your
> phone is then able to address packets to the real IP address on the
> other end, that end can't reach your phone through NAT.
>
> You may also have a codec selection problem. You should probably
> clean out the passwords from a copy of your sip.conf, and post it
> here. At the same time, check which codecs your phones support, and
> whether they can be set up with preferences for codec choice.
>
> -tih
> --
> Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
> www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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