[Asterisk-Users] trying to get SIP to work remotly.
Ross C
wotech at cox.net
Mon Dec 12 12:50:59 MST 2005
I had the same problem. It ended up being some settings in sip.conf
One of these settings did it for me (not sure which one, as I added them all
at once, then it worked):
Port=5060
localnet=192.168.1.0/255.255.255.0 ;<-----wutever the local subnet
is (that the asterisk server is on)
nat=yes
externip=68.92.31.19 ;<----wutever your public IP address is if you
asterisk server is behind a NAT firewall (don't worry, the one listed is
bogus; it's not my real addy)
this is in my sip_additional.conf file where 204 is the extension number of
the remote extension:
[204]
username=204
type=friend
secret=204
record_out=On-Demand
record_in=On-Demand
qualify=yes
nat=yes
insecure=very
mailbox=204 at default
host=dynamic
fromuser=204
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <204>
I'm guessing someone else will chime in to say that these settings aren't
correct for everyone, but this is just what worked for me. I have my
asterisk server behind a linksys wrt54g with the DMZ configured to go to the
asterisk server.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jason Brashear
Sent: Monday, December 12, 2005 1:23 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] trying to get SIP to work remotly.
I am working with Xten lite for now. I am able to register in but when I
call out
I can't hear anything. The caller on the other end can hear me just fine.
Any ideas?
I can get SIP to work fine internally.
I also have all the ports open in the firewall including 10000 - 200000
-J
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051212/8a2671a9/attachment.htm
More information about the asterisk-users
mailing list