romertron's profile

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2 Messages

Sun, Aug 17, 2014 7:30 PM

3rd Party VoIP Outgoing Call Issues - NVG589 Firmware Ugprade 9.1.0h12d22?

So I got a brand new Motorola NVG589 about a month ago to replace my 2Wire modem for my UVerse Internet service. I have VoIP with JoiPhone running through a Linksys RTP300. The RTP300 is directly connected to the NVG589. JoiPhone worked fine through a DMZ with the 2Wire.

 

I was initially able to get JoiPhone to work fine with the NVG589 by simply opening all of the ports with a custom service under Firewall -> NAT/Gaming. This was on 7/26/14.

 

JoiPhone Open All Ports for Motorola 7.26.14.JPG

 

I had no JoiPhone problems until 8/10/14. Then I started having one-way voice issues. I have a dialtone, I can call out, and the calls go through (I get missed message alerts on my cell), but I cannot hear anyone on the other end. So far I am able to hear when someone who calls me; the issue seems to be outgoing calls only.

 

I assigned the RTP300 a fixed IP address, and I placed it in IP Passthrough - DHCP-Fixed (I tried the Manual option, too). Same problem. I called JoiPhone and they did the same thing. They also tried to open the JoiPhone specific TCP/UDP ports in NAT/Gaming. Same problem. There were multiple power cylces of both the NVG589 and RTP300. JoiPhone recommended that the NVG589 be Bridged and/or that SIP ALG be disabled. I checked with AT&T and neither of these options are available for the NVG589.

 

I called AT&T and found out that I was pushed firmware version 9.1.0h12d22 on 8/6/14. It's certainly possible that this started the problems because it's unlikely that we used the phone between 8/6 and 8/10. Does anyone know what could have changed in the firmware update? Is there any way to roll it back, or is the prior one out there somewhere for troubleshooting?

 

I found this quote in another post: "The U-Verse service requires the customer to use a router that was designed specifically for U-Verse Internet, TV, and phone service.  This router has a default setting called SIP-ALG that cannot be disabled and the router cannot be replaced while using U-Verse Internet.  SIP-ALG is a router function that will cause VoIP traffic to be rewritten, which can cause problems such as one way audio, dropped calls, and inbound and outbound call failure.  If you are familiar with making changes to your voice over IP phone/device, you will want to modify the source port of the phone to a port other than 5060 and reboot the U-Verse router."

 

I don't have access to those settings in the RTP300, but I'm checking with JoiPhone to see if there's anything I can do. It still doesn't make sense that the phone worked for 3 weeks on source port 5060 and then started having issues unless there was a change in the firmware upgrade (or my RTP300 just flaked out--but then I wouldn't expect it to be working halfway).

 

Any other ideas to try?

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Contributor

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2 Messages

7 y ago

They were able to change the RTP300 source port from 5060 to 5066 remotely. I power cycled the NVG589 and the RTP300.

 

I already had the RTP300 MAC address in IP Passthrough DHCP-Fixed mode, but I was still having the same problem. However, when I went into NAT/Gaming and reopened the JoiPhone ports there, I noticed that the RTP300 activity on the Diagnostics -> NAT Table picked up. I tried the phone again, and voila, it works! This really doesn't make sense, because I thought the IP Passthrough mode should have been doing the same thing as the NAT/Gaming settings...? In fact, I just removed the RTP300 from IP Passthrough, and this didn't cause any issues.

 

Here are the open ports:

 

JoiPhone Ports.JPG

 

Anyway, it's working now and hopefully this solves the problem for good!

lottytx

Teacher

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73 Messages

@romertron

Very old post so chekcing to see if you still think this is necesary in most recent versions of software. I have my RG in passthru and turn off all packet filtering and the advanced settings.

So questin is do you need to add these specific settings and if so what are they doing relative to baseline.

My VoiP works fine but other side hears jitter and tunnel. Started late Dec 2019.

Thoughts?

spoom2

ACE - Master

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13.9K Messages

Since this post is 6 years old and in the wrong forum, I suggest you post your question in the Internet forum.

This is a user to user help forum. It is not an AT&T support site. I am not an AT&T employee, read the ACE's disclaimer at the bottom of this post

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