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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