Outcoming call... I don't hear

[koolyce]

Hi,

Mostly all the time now, the first time I dial a number from my FPL ATA, nothing happens on the line, but based on the person at the other side, he can hear me. So I recall and now, everything is fined.

My ATA is connected to my router, DLINK DIR-655. Based on the tech from FPL and the forum, I do port forwarding for the ports for UDP: 5060,5061,13000,13001. I also changed the firewall settings for the NAT endpoint filtering to Endpoint Independent.

But even with this config, I still have the problem and I don't know what to do next.

Anyone have suggestion?

Thanks

[dibsmft]

Try running STUN.

[koolyce]

Try running STUN.

Ok, I will try. Any better STUN server than another?

[dibsmft]

You need a server that is known to work such as stun.callwithus.com.

[cu2o2o2]

Hi,

Mostly all the time now, the first time I dial a number from my FPL ATA, nothing happens on the line, but based on the person at the other side, he can hear me. So I recall and now, everything is fined.

My ATA is connected to my router, DLINK DIR-655. Based on the tech from FPL and the forum, I do port forwarding for the ports for UDP: 5060,5061,13000,13001. I also changed the firewall settings for the NAT endpoint filtering to Endpoint Independent.

But even with this config, I still have the problem and I don't know what to do next.

Anyone have suggestion?

Thanks

The reason why you can not hear the other party is that the RTP ports (13000-13001) that you opened are NOT the ports that your ATA is configured to use. Your ATA if it is Linksys is expecting signals on ports 16384-16482 by default, but since they are closed, the incoming signals are just dropped and that is why you don't hear the incoming party.

You mentioned FPL ATA but you did not mention the model number. You must verify that the SIP and RTP ports that your FPL ATA are configured to, are the ones that you port forwarded in your router. Ex. Linksys ATA use RTP ports 16384-16482 and SIP ports 5060-5061 by default and must be open on your router (unless you changed them).

On my DIR-615 (your DIR-655 might be slightly different), I use port triggering and have to enable SIP ALG for it to work. Port forwarding and disabled SIP ALG also work (I must emphasize that SIP ALG must be disabled when port forwarding), but since I don't like my router ports always open, I am still using port triggering. DD-WRT also works, in fact, it works with VOIP without doing port forwards. I have tested these three setups to be equally working, however, YMMV as there could be other variables in play.

[koolyce]

The reason why you can not hear the other party is that the RTP ports (13000-13001) that you opened are NOT the ports that your ATA is configured to use. Your ATA if it is Linksys is expecting signals on ports 16384-16482 by default, but since they are closed, the incoming signals are just dropped and that is why you don't hear the incoming party.

You mentioned FPL ATA but you did not mention the model number. You must verify that the SIP and RTP ports that your FPL ATA are configured to, are the ones that you port forwarded in your router. Ex. Linksys ATA use RTP ports 16384-16482 and SIP ports 5060-5061 by default and must be open on your router (unless you changed them).

On my DIR-615 (your DIR-655 might be slightly different), I use port triggering and have to enable SIP ALG for it to work. Port forwarding and disabled SIP ALG also work (I must emphasize that SIP ALG must be disabled when port forwarding), but since I don't like my router ports always open, I am still using port triggering. DD-WRT also works, in fact, it works with VOIP without doing port forwards. I have tested these three setups to be equally working, however, YMMV as there could be other variables in play.

Hi,

I have a GrandStream-HT287 but I will look at my config.

I setup the STUN and seems to be better, but I will look for the port and also about your SIP ALG. Could you tell me more about SIP ALG because I didn't remember to see this in my router. In my case it should be disabled because I do port forwarding.

Thanks

[cu2o2o2]

Hi,

I have a GrandStream-HT287 but I will look at my config.

I setup the STUN and seems to be better, ...

If STUN works for you, then you should stick to it.

... but I will look for the port and also about your SIP ALG. Could you tell me more about SIP ALG because I didn't remember to see this in my router. In my case it should be disabled because I do port forwarding.
Thanks

SIP ALG settings for my DIR-615 is under the Firewall Settings of the Advanced tab, IIRC. Yours may be the same depending on the firmware.

See more about SIP ALG:
http://lmgtfy.com/?q=SIP+ALG

[dibsmft]

I don't think early firmware allowed the SIPALG to be diabled so you might have to upgrade your firware before you can turn it off.

[bridonca]

You are going to have to access your ATA, and find the SIP port, which should be 5060, and change it to 6060. You will also have to look got the RTP port, which I believe is 5004.

If you did not update the firmware on your DIR-655, I strongly recommend you do so. The default firmware in the DIR-655 has been known to cause problems with SIP VOIP, and the newer firmware has a fix.

The reason I am getting you to change the SIP port to 6060 is so you can easily bypass any potential SIP ALG issue, which only affects port 5060. Just make sure you set the DIR-655 to route ports 6060 and 5004(or whatever the RTP port is) to the ATA.

[koolyce]

You are going to have to access your ATA, and find the SIP port, which should be 5060, and change it to 6060. You will also have to look got the RTP port, which I believe is 5004.

If you did not update the firmware on your DIR-655, I strongly recommend you do so. The default firmware in the DIR-655 has been known to cause problems with SIP VOIP, and the newer firmware has a fix.

The reason I am getting you to change the SIP port to 6060 is so you can easily bypass any potential SIP ALG issue, which only affects port 5060. Just make sure you set the DIR-655 to route ports 6060 and 5004(or whatever the RTP port is) to the ATA.

Thanks for you advice. I will also try this by changing the SIP port to 6060.

What I see tonight is that the option "Random port" just below SIP and RTP port is to YES. So I changed this to NO.

I also try to deactivate the SIP ALG in my router..ERROR... no more phone so I need to keep it there but with the hint of bridonca, I hope this will be good.

I still use the STUN but I don't use this option: "Use STUN keep-alive to detect networks connectivity", should I?

I will keep update in the next few days to inform you what are my results.

Thanks again

[dibsmft]

Using port 6060 UDP for the Grandstream SIP should be just fine as long as you have it in the ATA configuration .... it should make no difference but it is not a bad idea nor to use 5060 UDP here.
I have "Random port" set to NO as well. It is OK to use YES if you have plenty of open ports and have not done port forwarding.
If things work without deactivating SipAlg then be thankful and leave it that way ... I think forwarding ports is all you will need.
I have the keep-alive on and set..... keep-alive interval: 20 (in seconds, default 20 seconds)

We await the results of your testing. Good luck!

[koolyce]

The results seems to be there and I don't have (for the moment) anymore the problem for the outcoming call.

There is one problem left but I don't know if it is from me or no. When my stepmother call us, she told us that they ear some "click" during the call. But when we call her, no "click". Anyone have an idea about that? She told us that the "click" is only when she call us.

Thanks for the help.

[TheHardy]

For the most part, I cannot overstate how important it is to always make sure that your firmware is the best possible for your application. Latest firmware updates commonly fix issues, and specific firmware upgrades that are known to work (rather than latest ones) are sometimes needed too.

[koolyce]

For the most part, I cannot overstate how important it is to always make sure that your firmware is the best possible for your application. Latest firmware updates commonly fix issues, and specific firmware upgrades that are known to work (rather than latest ones) are sometimes needed too.

How can we update the firmware for the GrandSteam HT-287? I didn't see anything about this on the website.

[TheHardy]

Not sure that there is a firmware upgrade for the HT-287 ATA, I was more commenting on ROUTERS, which notoriously ship with outdated firmware. The ATA's should have a decent enough firmware load out of the box, as they are DESIGNED for VOIP, whereas most routers are just simple network devices which get used for VOIP ...