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
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