Grandstream HT-701

[befernandes]

Can somebody please help....I just bought this from Amazon and I cant seem to even get to the login page. Everytime I hit the **** on the phone it says the ip is 0.0.0.0 and when I put in it to my browser I get nothing. What am i doing wrong?

[Bloodsong]

0.0.0.0 is a non-routable IP address.
It is basically the phone saying the ATA has not received an IP address on your network.

Let's go with the basics first,
Test a new Ethernet cord from your router/switch to your ATA.
Ensure that DHCP is enabled over the wired LAN from one of your devices.
The "Link/Act" Light should be on, green, and preferably flashing somewhat.

I haven't looked at the product sheet for the HT-701, but generally there is a solid green light if you have a fully negotiated link, and it will blink occasionally (or rapidly) depending on the amount of data being transferred.

I see there is also an "Internet" light, as this is a network device and not a switch/router itself this is also important to watch for and will denote it's connectivity state at a layer 3 level. (As opposed to layer 2 is is link/act) This means IP address and the general ability to be routed to the internet.

If none of these come-on, and you're certain you have a good cable, and you're certain you have DHCP enabled on your router for the wired LAN, then your next step is to hit the reset button on the HT-701.

RESET BUTTON
Reset default factory settings following these four (4)
steps:
1. Unplug the Ethernet cable.
2. Locate a needle sized hole on the back panel of the gateway unit next to the power connection
3. Insert a pin in this hole, and press for about 7 seconds.
4. Take out the pin. All unit settings are restored to factory settings

Give it about a minute after completing this, plug it back into your network, give it some time to get an address, and dial the IVR again.

There are some other steps we can try, but would like to see the results of this first.

[befernandes]

Thanks! got it working late last night. You are right I had it connected after a modem, through a wireless router to a switch then the ATA. Once I plugged it straight to the wireless router I got the ip and was able to login. BUT for some reason now I am unable to receive calls. I can only make calls see pic: http://easycaptures.com/fs/uploaded/712/8996639038.png any thoughts?

Update: I just unplugged and restarted both the wireless router and the ATA and it started working again (receiving calls) BUT stopped receiving calls after about half an hour. Its definately not the wireless router, instead a setting on the grandstream 701 which is making it time out. When I check the status of the device on the FPL website it says:

SIP Status: disconnected
SIP User Agent:

[Trigger]

Is there any update to the config settings for the Grandstream ht-702 with the latest firmware update (1.0.5.10)? I'm able to get it working except for when someone calls, I can't hear them, they can hear me though. I switch the Nat transversal to pnup as mentioned in a previous post to no avail.
There seems to be a lot of conflicting info for this ata on the freephoneline system, so I'm wondering if there is something current and working.

Thank you.

[kpsingh]

Dear Friends,

May I request you please if you can share Grandstream HT-702 ATA's working configuration with Freephoneline ( Not with Fongo ) . I am trying to configure this ATA, outgoing calls are working fine but sometimes for the incoming calls other party is getting a Not available message and calls land into Voicemail. I have tried uPnP, Keep Alive but the problem was not solved. I tried STUN then other party is not able to hear ring in their headpiece and call suddenly connects.

Thank you very much in advance.

Regards,
KP

[bridonca]

May I request you please if you can share Grandstream HT-702 ATA's working configuration with Freephoneline ( Not with Fongo ) . I am trying to configure this ATA, outgoing calls are working fine but sometimes for the incoming calls other party is getting a Not available message and calls land into Voicemail. I have tried uPnP, Keep Alive but the problem was not solved. I tried STUN then other party is not able to hear ring in their headpiece and call suddenly connects.KP

Your appear to have a router issue. Your HT-702 need an available RTP port open forwarded from the router to the ATA. The default RTP port the HT-702 uses is 5004.

What do you have for a router?

[kpsingh]

May I request you please if you can share Grandstream HT-702 ATA's working configuration with Freephoneline ( Not with Fongo ) . I am trying to configure this ATA, outgoing calls are working fine but sometimes for the incoming calls other party is getting a Not available message and calls land into Voicemail. I have tried uPnP, Keep Alive but the problem was not solved. I tried STUN then other party is not able to hear ring in their headpiece and call suddenly connects.KP

Your appear to have a router issue. Your HT-702 need an available RTP port open forwarded from the router to the ATA. The default RTP port the HT-702 uses is 5004.

What do you have for a router?

@bridonca,

Thanks for the reply. This problem does not happens all the time. It happens only sometimes for the incoming calls. Other party is getting a Not available message and calls land into Voicemail.

I have a PAP2TNA and I have two lines configured on that and both of them are working fine behind the same router. I was trying to configure HT-702 and this problem was happening. I thought the reason of this problem might be due to multiple SIP devices working behind one router causing this problem, so I configured HT-702 and gave it to my friend.
3-4 days later my friend told me that he is facing this issue. He has a Rogers Modem+Router combo device ( DPC3825 ). After Restarting/Rebooting HT-702 this problem gets solved. But this problem re-appears after some time. So it looks like Registration issue ? or Keep Alive / NAT stuff not working properly. That is why I was thinking if I can get working configuration of HT-702 , it might be easy for me to pin point this issue.

Earlier I had set-up it as
Use Random SIP Port: Yes
Use Random RTP Port: Yes

Yesterday I have changed that to following, so let's see if this problem gets fixed.

Use Random SIP Port: No
Use Random RTP Port: No


My Home Configuration :
Teksavvy ISP -- > Thomson Cable Modem --> Linksys WiFi Router WRTU54G-TM --> [ PAP2T-NA + HT-702 ]

My Friends Configuration:
Rogers ISP -- > DPC3825 --> [ HT-702 ]

[bridonca]

Are you using one FPL account for 2 ATA devices? If so, that is your problem. Your freephoneline account has 2 outgoing trunks and only 1 incoming line. So if you have 2 ATA devices running the same account, you can make 2 separate outgoing phone calls just fine, but the incoming call will be flaky, because it is split between two ATA devices. This was set up to allow for call waiting to work, not for 2 separate lines.

Assuming the above is not the issue, I strongly believe you are getting flaky results because you have a router with flaky firmware. Fortunately, there is an available OpenWRT firmware for your WRTU54G-TM. I strongly recommend you install it. http://wiki.scottn.us/wrtu54g-tm

With this flaky WRTU54G-TM firmware, I am shocked the PAP2 works fine, and the HT-702 is causing you problems. Usually the PAP2 is the prima donna, and the HT-702 just works.

Still there as a potential for conflict. It is never good to allow the router to figure it out. You should map ports for each device.

By default the PAP2 wants RTP ports 16384 to 16482, so get the router to forward these to the PAP2. As already stated, the default RTP port for the HT-702 is 5004, so get the router to forward 5004 to the HT-702. Make sure each respective ATA is set up to accept those ports. Though it should not matter, for good measure, Map port 6060 as a SIP port for the HT-702, and make sure the HT-702 and router are set up to accept 6060 as a sip port. After you save settings, and power down of the router ATA and modem, see what happens.

[kpsingh]

Are you using one FPL account for 2 ATA devices? If so, that is your problem. Your freephoneline account has 2 outgoing trunks and only 1 incoming line. So if you have 2 ATA devices running the same account, you can make 2 separate outgoing phone calls just fine, but the incoming call will be flaky, because it is split between two ATA devices. This was set up to allow for call waiting to work, not for 2 separate lines.

Assuming the above is not the issue, I strongly believe you are getting flaky results because you have a router with flaky firmware. Fortunately, there is an available OpenWRT firmware for your WRTU54G-TM. I strongly recommend you install it. http://wiki.scottn.us/wrtu54g-tm

With this flaky WRTU54G-TM firmware, I am shocked the PAP2 works fine, and the HT-702 is causing you problems. Usually the PAP2 is the prima donna, and the HT-702 just works.

Still there as a potential for conflict. It is never good to allow the router to figure it out. You should map ports for each device.

By default the PAP2 wants RTP ports 16384 to 16482, so get the router to forward these to the PAP2. As already stated, the default RTP port for the HT-702 is 5004, so get the router to forward 5004 to the HT-702. Make sure each respective ATA is set up to accept those ports. Though it should not matter, for good measure, Map port 6060 as a SIP port for the HT-702, and make sure the HT-702 and router are set up to accept 6060 as a sip port. After you save settings, and power down of the router ATA and modem, see what happens.

@bridonca

Thanks once again for your reply. My friend got his separate Freephoneline SIP Account/Password. I was not re-using my account on his ATA.
As per your suggestion today I have put my Friends HT-702 in DMZ mode behind his router DPC3825, Let's see if the problem is resolved by this fix. I will keep you posted.

My Friends Latest Configuration:
Rogers ISP -- > DPC3825 --> [ HT-702 assigned static local IP + Google's Public DNS and On DMZ ]

[kpsingh]

Are you using one FPL account for 2 ATA devices? If so, that is your problem. Your freephoneline account has 2 outgoing trunks and only 1 incoming line. So if you have 2 ATA devices running the same account, you can make 2 separate outgoing phone calls just fine, but the incoming call will be flaky, because it is split between two ATA devices. This was set up to allow for call waiting to work, not for 2 separate lines.

Assuming the above is not the issue, I strongly believe you are getting flaky results because you have a router with flaky firmware. Fortunately, there is an available OpenWRT firmware for your WRTU54G-TM. I strongly recommend you install it. http://wiki.scottn.us/wrtu54g-tm

With this flaky WRTU54G-TM firmware, I am shocked the PAP2 works fine, and the HT-702 is causing you problems. Usually the PAP2 is the prima donna, and the HT-702 just works.

Still there as a potential for conflict. It is never good to allow the router to figure it out. You should map ports for each device.

By default the PAP2 wants RTP ports 16384 to 16482, so get the router to forward these to the PAP2. As already stated, the default RTP port for the HT-702 is 5004, so get the router to forward 5004 to the HT-702. Make sure each respective ATA is set up to accept those ports. Though it should not matter, for good measure, Map port 6060 as a SIP port for the HT-702, and make sure the HT-702 and router are set up to accept 6060 as a sip port. After you save settings, and power down of the router ATA and modem, see what happens.

@bridonca

Thanks once again for your reply. My friend got his separate Freephoneline SIP Account/Password. I was not re-using my account on his ATA.
As per your suggestion today I have put my Friends HT-702 in DMZ mode behind his router DPC3825, Let's see if the problem is resolved by this fix. I will keep you posted.

My Friends Latest Configuration:
Rogers ISP -- > DPC3825 --> [ HT-702 assigned static local IP + Google's Public DNS and On DMZ ]

I have put my Friend's HT-702 in DMZ mode behind his router DPC3825 , and this has resolved all of the problems.

Another issue we noticed that his Number was ported from Previous VOIP provider to Freephoneline.ca, after porting we thought that the previous provider would automatically cancel his number on their network but interestingly this was not the case and his number was active on previous Voip Provider as well as simultaneously on Freephoneline.ca. Because of this following scenario was encountered by us ( Because same number was active on two VIOP providers network ) :

Other VOIP customers on Previous VOIP's network ---- > Were going into his Voice mail on old VOIP network ( we unplugged / power-off his old ATA )
Other Telecom/Voip customers ----->>> Could connect to his new Freephoneline.ca ( New ATA configured with Freephoneline.ca account and switched on )
Other VOIP customers on Previous VOIP's network ---- > Could connect to his Old Viop account ( when we plugged in / switch on his old ATA )

In Nutshell while porting out to other VOIP provider , it seems the previous VOIP provider is not cancelling your number on their network . So after porting is completed we need to call previous VOIP provider and cancel our number from their network. It seems home network of previous VOIP provider was not routing this call correctly out of their network , their home network was confused as the number entry was present in their home network number database.