It's a Cisco ASA 5510. I have now disabled port randomization for the NAT, 
looks like the media control ports 30000-31000 are pretty important and need to 
stay the same. I attempted to turn off the NAT functions within sipX (based on 
an archived post you made) but it appears to break DTMF on the AA with my 
setup. I've also shut MOH off in sipXbridge for now, we can deal with the 
silence. It looks like by default that the moh uri for the Polycom's is blank. 
Thanks for the help.

-----Original Message-----
From: Tony Graziano <tgrazi...@myitdepartment.net>
Date: Wed, 6 Jan 2010 21:06:03 
To: <nathan.nieb...@sacatech.com>
Cc: <sipx-users@list.sipfoundry.org>
Subject: Re: [sipx-users] Brain picking

Then you need to disable the alg on your firewall.

Your firewall is the issue. What firewall is it? I did a hotwo on pfsesne
which explains in plain words what sipxbridge is looking for from a
firewall. The firewall needs to do symmetric port nat, no application layer
gateway (alg), or you need to use another sbc, especially for trunking. We
install ingates for that.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: tgrazi...@myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: nathan.nieb...@sacatech.com <nathan.nieb...@sacatech.com>
To: Tony Graziano <tgrazi...@myitdepartment.net>
Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org>
Sent: Wed Jan 06 20:36:59 2010
Subject: Re: [sipx-users] Brain picking

I have been all over the wiki. Pieced a bunch of stuff together, none seem
to fit my scenario just right.

For example, I am delivering calls from bandwidth.com to sipXbridge on port
5080. sipX is sitting with a private ip behind a sip aware firewall. My
phones can communicate with sipX without restriction over our WAN. So I am
wondering if I even need to have the trunk pointing at 5080 rather than
5060?

We are having issues with call park where I will attempt to pickup a call,
the caller can hear the hold music stop for a split second, and then the
call stays parked or gets disconnected.

Randomly we will try to answer a call and its like the "off hook" signaling
is not sent.

Also, I can put a call on hold but when we go back to pick it up the audio
stream becomes 1-way (I can hear the caller but they cannot hear me).

I know enough to know any RTP stream is given free reign between sipX and
any registered phone so these issues should not be occurring unless its
something between the bridge, the proxy or the provider. My default codec is
also set to g.729.

Cisco firmware I've used are 8.2.2sr4 and 8.5.3 and Polycom 3.1.3c.

------Original Message------
From: Tony Graziano
To: nathan.nieb...@sacatech.com
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Brain picking
Sent: Jan 6, 2010 4:24 PM

Why don't you use the howto's I posted?
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: tgrazi...@myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: sipx-users-boun...@list.sipfoundry.org
<sipx-users-boun...@list.sipfoundry.org>
Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org>
Sent: Wed Jan 06 19:21:32 2010
Subject: [sipx-users] Brain picking

Anyone willing to let me pick their brain so I can get my setup working
100%?

Thanks

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