VOIP phone connection problem

I have been using a Verizon dry loop DSL and VOIP (VYLMedia)  for my phone for about two years.   About one year ago (with no equipment change) the call would temporarily drop and then reconnect.   This was a hindrance but at least you could talk.  The past couple of months the frequency of dropping out during a call as about every 3 sec and I cannot carry on a conversation this way.   I was told by my VOIP provider, 1-VOIP, that this is called jitter.  
I have been working with the tech rep for over a week now and we cannot get it resolved.   I have a Westell Versalink 327W wireless router (Verizon) and a Linksys SPA2102-R phone adaptor from 1-VOIP.  He did alot of tweaking to no avail.   He sent me a new phone adaptor this morning and that didn't help.   
After this morning's tweaking session this is what he told me:
You are currently being registered on 50484 to our system, which can indicate that your ISP and/or equipment is blocking the proper authentication and registration on the appropriate ports.
You will need to forward the following ports to the phone adapter in order to have consistent phone quality as well as incoming phone calls:
5060-5160 UDP
10,000 – 20,000 UDP
You can still obtain this configuration with the proper ports forward which will ensure you have consistent and proper voice quality.
Using a link I found in this forum I tested the quality of line for speed and jitter.   It showed 0.0mg jitter and no packet loss.  The MOS score was 3.9.   Upload:  738Download:  2.8
I am now thoroughly confused and have no idea how to get this resolved.   Any takers?

Obviously the line is transferring data as it should. Do the following:
Log into the Westell at http://192.168.1.1/ using one of the following login credentials:
admin/password
admin/password1
admin/admin
admin/admin1
From there, go to System Monitoring, click on Advanced Monitors, and then choose Transceiver Statistics. From there, close the Transceiver Statistics page and then visit the System Logging section. Select the "All" log and post up the contents of that log. If you get an error about the post being too lengthy, use http://pastebin.com/ to post the log by providing the URL of your paste.
Next, go to the Firewall section of the Westell and ensure the Firewall is not by any chance enabled. If it is enabled, disable it by selecting "No Security."
========
The first to bring me 1Gbps Fiber for $30/m wins!

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    On the H323 gateway I have installed one BRI interface and one FXS interface.
    Inside my network I am using VoIP telephones 7910.
    Originating calls in the Voip telephones which destination is the PSTN is working fine;
    Receiving calls from PSTN that destination is VoIP telephones is OK also;
    Redirecting or doing “Hold” on calls between VoiP telephones works but…..
    When redirecting or doing “Hold” on calls between a VoiP telephone and a call from PSTN I loose the voice stream.
    Apparently, all signalling is working fine because when redirecting the call originated in the PSTN from one VoIP telephone to another VoIP telephone, I can see the Calling number ID. Even if I use the telephone keyboard, I can listen the tones, but I have no voice stream.
    Please any help will be welcome.
    Thanks,
    David Costa

    I have tried different configurations and now it is working. The problem is that I couldn’t identify were the miss configuration was.
    Another question is: Can I use a codec with compression between voip interface of h323 gateway and the Callmanager or Voip phones to pass music on hold?
    Now I only pass music on hold if I configure G.711 on the voip interface of h323 gateway.
    Thanks,
    David Costa

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