Karl Brose wrote:
I think when you have this setup you need to keep the media path going
through Asterisk at all times.
Your SIP is binding to both ports, internal and external, but that
doesn't correctly set it up for either scenario, localnet calls and
external calls. It won't keep the addresses straight for the RTP
channels.
Try setting CANREINVITE=NO for peers (FWD,..) and for your local SIP
phones. When a channel is created in asterisk the media path is going
through Asterisk, but during a call the endpoints can issue reinvites
which switches the media path directly between the endpoints. You need
to prevent that.
Other solutions are to run IAX to/from FWD and SIP locally, or SIP to
the external peers and IAX to a local IAX phone (or another protocol).
Or you should be able to create your own NAT using the iptables and
bind asterisk only on one port either outside or inside and set the
right corresponding parameters. The RTP will still bind on all ports
currently, but that will be fixed in a matter of days.
Also, sipgate.net should be sipgate.de (works ok though since they
don't care)
fromdomain is meant to be realm not a hostname.
Thomas Gallaway wrote:
Hi all
I use sipgate and FWD but seem not to get it going. I do not have NAT
on the asterisk box (static ip).
The asterisk box has 2 network interfaces. One internal and one
external.
Now when I make an call to a FWD or SipGate number all I get is
-- Executing NoOp("SIP/113-6d2e", "") in new stack
-- Executing Goto("SIP/113-6d2e", "intern-post|714551|1") in new
stack
-- Goto (intern-post,714551,1)
-- Executing SetCallerID("SIP/113-6d2e", "270002") in new stack
-- Executing SetCIDName("SIP/113-6d2e", "Thomas Gallaway") in new
stack
-- Executing Dial("SIP/113-6d2e", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fwd270002-6ee7 answered SIP/113-6d2e
-- Attempting native bridge of SIP/113-6d2e and SIP/fwd270002-6ee7
== Spawn extension (intern-post, 714551, 3) exited non-zero on
'SIP/113-6d2e'
But either I get 1/2 second of audio or no audio. No matter how long
I wait there is just no audio or just a short snippet of audio at the
beginning.
Here is parts of my sip.conf;
[general]
port = 5060 ; Port to bind to
localnet = 192.168.1.0 ; Internal NETWORK address
localmask = 255.255.255.0 ; Internal netmask
externip = 206.40.161.235
context = intern ; Default for incoming calls
maxexpirey=3600
defaultexpirey=300
disallow=all ; Disallow all codecsa
allow=gsm
allow=alaw
allow=ulaw
tos=reliability
register => xxx:[EMAIL PROTECTED]/150
register => xxx:[EMAIL PROTECTED]/151
[sipgate1]
type=friend
username=xxx
secret=xxx
host=sipgate.de
fromuser=xxx
fromdomain=sipgate.net
nat=no
context=incoming-sipgate
canreinvite=yes
[fwd270002]
allow=ulaw
type=friend
context=incoming-fwd
secret=xxx
username=xxx
host=fwd.pulver.com
Any ideas?
When I put nat=yes I actually will get 1 second of audio, then it dies.
I have been googling for a while now and not seem to find any
sollution to this.
-- Thomas
I will try that. I had to remove all the IAX / SIP changes I did as even
on the local network it started to give me a one way communication
thing. I was able to hear other people but they could not hear me.
Will give this another try later in the week.
-- Thomas
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