Thanks to all who replied.  Problem is solved.
 
Deleted my custom dialing plan, and went back to the Local and Long Distance 
dialing plans that are built in.
 
No firewall changes were made.  I believe it was 100% voipvoip at this point.  
Basically I set up the new trunk to voip.ms and went back to using two of the 
default dial plans.
 
I now dial 9, then the 11 digits...1+area code+phone number.  Worked the first 
time to my cell phone.  I then called a relative out of state and that worked 
the first time.  Audio worked both directions in both tests.
 
On Cisco firewall appliances and blades that are running the version 8 IOS 
(which is Cisco's unified and go-forward platform), the fixup (and all previous 
PIX nomenclature) is gone.  It's deprecated and they are no longer supporting 
PIX--I believe you can't even buy SmartNet for that platform any longer.  
Someone still using it will have to verify/correct me on that.  We ditched PIX 
3+ years ago.
 
We have Cisco ASA appliances running version 8.2.2.  We have a 1-to-1 NAT set 
up so that the SipXecs server is the only one that uses the associated public 
IP address.  We allow ALL outgoing ports, none incoming--there is nothing from 
the public internet allowed to initiate any connections to the SipXecs server.  
For a SIP trunk where our side initiates it, you shouldn't need any incoming 
ports--the ASA is stateful by default for both TCP and UDP.
 
Additionally, we have TCP sequence randomization turned off for our 1-to-1 NAT, 
as it was recommended in more than one document I read to do this from the 
outset.
 
I will be proceeding to test the functionality of the system now.  I appreciate 
all the assistance.  I hope to be able to contribute back to the list some kind 
of way in the future.


--- On Wed, 12/8/10, Tony Graziano <[email protected]> wrote:


From: Tony Graziano <[email protected]>
Subject: Re: [sipx-users] Can't place outgoing calls - fast busy
To: "Discussion list for users of sipXecs software" 
<[email protected]>
Date: Wednesday, December 8, 2010, 5:01 PM



When sipx sends to voip.ms, it does so on port 5060. When you register with 
voip.ms you register on port 5080, and they send inbound calls to port 5080 
(following the registration). You can view the ip:port your registered from at 
their portal. if ti does not show the public ip of your firewall and port 5080, 
then there's your message being thrown.


So you need rules for both 5060/5080.


what firewall are you using? its not pix, what is it?

On Wed, Dec 8, 2010 at 5:49 PM, Victor Williams <[email protected]> wrote:






Doesn't apply to our setup...we're not using pix firewalls, and that command 
set has been deprecated--like 3 years ago.  There is no inspection of packets 
on tcp/udp ports 5060 or 5080 going on.
 
I would expect some log file somewhere to be throwing some message that 
indicates what's going.
 
No one has answered my question previously;  if there's a port 5080 issue, how 
can an incoming call work flawlessly and outgoing not work at all?  Wouldn't 
that affect BOTH directions?
 
Turning up the logging at the firewall indicates exactly what we should be 
seeing...connections established and torn down like they're supposed to be.  
When I attempt a call out, there is no connection even attempted.  This is why 
I'm leaning towards a configuration issue on the side of SipXecs...which I am 
not familiar with really.  I'm still asking for help in where to look.  
Currently the sipxbridge.log file has been helpful with registration issues and 
looking at what happens when a call comes in.  However, I would expect some 
entries in that file when a call goes out as well.  I'm seeing nothing written 
to that file when I attempt to dial out.
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-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
Fax: 434.326.5325

Email: [email protected]

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]

Helpdesk Contract Customers:
http://support.myitdepartment.net


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