FPL with HT-701 (Firmware 1.0.8.2)

[mh1983]

Hi all,

I've collected these items into a document over the months. The doc builds upon the great info from Jake/Val and Mango in the past and is working config for 1.0.8.2 firmware on HT-701.

I've gotten a lot of help on these forums since I started using FPL. I wanted to share this out to pay that back, even in a small way. Hope this will help someone get up and running.


Edit: Disable Allow Incoming SIP Messages from SIP Proxy Only". Change that to "No". FPL to FPL and Fongo to FPL calls may not work with that setting enabled. Fongo Home Phone has that setting disabled.

[Fongo Support]

Hi,

Thank you so much for your efforts. I made your post a sticky.

Thanks again!

[Natalie]

I recently updated my firmware from 1.0.7 to 1.0.8.2; the only difference I noticed between my past settings to the ones listed in this guide are the change in SIP Registration Failure Retry Wait Time from 20 to 120. My form has 20 listed as a default. Just noticed today that when I tried to call out I got a "device not registered" message. Could this be due to this change?

[Liptonbrisk]

I recently updated my firmware from 1.0.7 to 1.0.8.2; the only difference I noticed between my past settings to the ones listed in this guide are the change in SIP Registration Failure Retry Wait Time from 20 to 120. My form has 20 listed as a default. Just noticed today that when I tried to call out I got a "device not registered" message. Could this be due to this change?

No. 120 is correct.
http://support.freephoneline.ca/hc/en-u ... redentials

[mh1983]

I recently updated my firmware from 1.0.7 to 1.0.8.2; the only difference I noticed between my past settings to the ones listed in this guide are the change in SIP Registration Failure Retry Wait Time from 20 to 120. My form has 20 listed as a default. Just noticed today that when I tried to call out I got a "device not registered" message. Could this be due to this change?

Also, for 1.0.8.2 -- Enable SIP Options Keep Alive: Yes

[leonnoel]

Hi. With these current settings, I'm not able to receive audio on incoming calls. However, audio is successfully transmitted on incoming calls, and audio is successfully transmitted and received on outgoing calls. Any help would be greatly appreciated. I'm using firmware 1.0.8.2 on the Grandstream HT-701.

Thanks

[Natalie]

I recently updated my firmware from 1.0.7 to 1.0.8.2; the only difference I noticed between my past settings to the ones listed in this guide are the change in SIP Registration Failure Retry Wait Time from 20 to 120. My form has 20 listed as a default. Just noticed today that when I tried to call out I got a "device not registered" message. Could this be due to this change?

Also, for 1.0.8.2 -- Enable SIP Options Keep Alive: Yes

Have all the items checked off as indicated in the Config guide including your suggestion above, and it still says device not registered about once a day.

[mh1983]

To the folks above having connectivity issues using HT-701 on 1.0.8.2:

Can you try changing this setting to the suggested value below?

SIP REGISTER Contact Header Uses: WAN Address

Are you using voip4.freephoneline.ca:6060? If not, I'd suggest to try it.

Also, check the suggestions here: viewtopic.php?f=8&t=18908&p=74214&hilit=bridge+mode#p74214

[mh1983]

Another thing to try as you work through some troubleshooting steps: consider changing the two DNS servers from the ISP-provided values to OpenDNS (what I use) or Google DNS values

Post back here with how it goes and I'll happily make changes to the doc if they're needed. Good luck!.

[mh1983]

v2 of the doc contains troubleshooting information with credit to Lipton and Mango.

[Liptonbrisk]

For those that are interested, Asus Merlin only works with the routers listed at https://www.asuswrt-merlin.net/about.

[mh1983]

v3 of the doc contains a setting I recently learned needs to be changed from the default value

SIP OPTIONS Keep Alive Interval -- change to 20 (30 is the default)

Thanks to Lipton for pointing it out in another thread.

It also contains the link to the supported Asus routers for Merlin firmware, and a procedure for upgrading the HT701 to 1.0.8.2 firmware.

[GenChick]

Thank you mh1983, for all hard work you have done: collecting info, posting and updating this guide! It is easy to follow and very helpful!

[mh1983]

Does anyone know how to disable or change the music on hold for the HT701? Mine has some really weird calypso music. It's pretty funny but awkward. I can't find any documentation or posts anywhere about this. Thanks in advance!

[Liptonbrisk]

Does anyone know how to disable or change the music on hold for the HT701? Mine has some really weird calypso music. It's pretty funny but awkward. I can't find any documentation or posts anywhere about this. Thanks in advance!

The source of the music is actually coming from Fongo/FPL.

I don't know of a way to disable it on Grandstream ATAs, but I've also never owned one.

[mh1983]

For 1.0.7.2 (I think), Mango called out this setting:

Once the new firmware is installed, set SIP REGISTER Contact Header Uses to WAN address.

I don't have this called out in the 1.0.8.2 guide, but I see that that field is present in the firmware -- the default is LAN address.

Does anyone know which is the preferred setting? I'd like to capture that in an update to the guide. Thx!

[Liptonbrisk]

For 1.0.7.2 (I think), Mango called out this setting:

Once the new firmware is installed, set SIP REGISTER Contact Header Uses to WAN address.

I don't have this called out in the 1.0.8.2 guide, but I see that that field is present in the firmware -- the default is LAN address.

Does anyone know which is the preferred setting? I'd like to capture that in an update to the guide. Thx!

Set that to WAN

[mh1983]

V4 w/ a minor update:
* Added "SIP REGISTER Contact Header Uses WAN Address" (thx, Mango and Lipton)
* Removed duplicate field "Transfer on Conference Hangup"

[Liptonbrisk]

"Ensure whatever modem/router combo your ISP gave you is in bridge mode" should be changed to "If you're using your own router, ensure whatever modem/router combo your ISP gave you is in bridge mode if you encounter problems, such as one-way audio or device registration issues."

If you don't have your own router, sticking a gateway or modem/router combo into bridge mode will leave your ATA unprotected.

[mh1983]

Thanks, Lipton! Here's v5 of the guide with the change you suggested.

[mh1983]

I'd like to update the original post with the latest guide but can't figure out how. Any pointers are appreciated!

[Liptonbrisk]

I'd like to update the original post with the latest guide but can't figure out how. Any pointers are appreciated!

Done. I deleted the old ones so that no one can accidentally download them.

[mh1983]

Thanks, Lipton -- appreciate it!

[mh1983]

I see that 1.0.10.3 was released for the 701 and 702 on May 29th (http://firmware.grandstream.com/Release ... 0.10.3.pdf). I upgraded my ATA and so far things seem to be working fine with the config I used for 1.0.8.2.

I will see how my ATA does over the next few days and report back/update the documentation as needed.

[mh1983]

So far so good. SIP OPTIONS Keep Alive Interval seems to have changed to a default of 30, whereas my guide calls out 20. That's the only difference I've noticed so far. Will monitor a little more before updating the guide.