Can't hear the caller when they call

[TKS]

I am using Grandstream HT701 and in its settings (FXS PORT) you need to check:

Check SIP User ID for incoming INVITE: NO
Allow Incoming SIP Messages from SIP Proxy Only: NO

This is contrary to what this board is saying, because when another IP phone or mobile phone (which is IP) is calling, I can't hear the caller. But after I changed the settings it works. But I had to also do a hard-reboot, not just a soft reboot through a browser.

[dibsmft]

Read on a different forum of someone having the same problem, a user recommended "NAT Mapping Enable" & "NAT Keep Alive Enable" to be turned on - tried it, still no progress. Not a call today, problem continues

You need a STUN server as well for your problem. Run STUN with a server such as stun.callwithus.com.

[vladshab13]

It seems I have found the problem. I have opened ports 5060,5061, 13000 and 13001 as UDP in my router and now everything is working.

[gershe]

For those using PAP2 adapter:

Goto: Line 1 tab and make sure following settings are set as follows:
NAT Mapping Enable: YES NAT Keep Alive Enable: YES
NAT Keep Alive Msg: $NOTIFY NAT Keep Alive Dest: $PROXY

Goto: SIP Tab (bottom):
Handle VIA received: YES Handle VIA rport: YES
Insert VIA received: YES Insert VIA rport: YES

Let me know if it helps please.

[robintre]

UDP ports 5060,5061,13000 and 13001 was already opened for me, but the problem started early this morning, worked fine for months before.

I finally found that the setting "NAT Traversal" was set to "Yes, STUN server is:" with server name set to blank (look into ADVANCED SETTINGS 1 of the Grandstream config page).
I set this to "No" and the problem is gone, I can hear on incoming call now.

Not sure what's STUN is used for, anyone can explain?

[dibsmft]

As far as I know router DMZ only makes those ports not already in use available so perhaps you have something else using ports that is clashing with you ATA. Lots of people have the same problem (or similar) to what you have. For the 3102 you probably need to run STUN, NAT keep alive, NAT mapping enable with a suitable STUN server. This should not be needed if you have forwarded ports but seems to be necessary for the Linksys devices (see manual).

[vladshab13]

gershe
Thanks. I have change settings, you wrote. Still working. Will see.

[iDrone]

I finally found that the setting "NAT Traversal" was set to "Yes, STUN server is:" with server name set to blank (look into ADVANCED SETTINGS 1 of the Grandstream config page).
I set this to "No" and the problem is gone, I can hear on incoming call now.

Thanks a ton robintre, this has helped for me too. Not sure if it is just temporary, but so far so good!

[dibsmft]

Check this link
http://forum.fongo.com/viewtopic.php?f=8&t=7348
Most of this is in the Sipura (Linksys manual)
If that is not enough the run STUN with a suitable stun server.

[TKS]

I originally posted changes to 2 SIP variables (SIP usr ID for INVITE = No, Allow SIP messages from SIP proxy = No), but that alone didn't work. I still experienced this problem.

After reading some more posts, I made this change in Grandstream configuration:
(on page 4 - FXS PORT) NAT Traversal (STUN) = changed to NO
(on page 3 - Advanced) STUN server is = <blank>

I think some users recommend to use STUN, but my problem was that is was mixed up, NAT Traversal (STUN) was set originally to YES, but there was no server.

It is working now for 6th phone call that I received, I could hear the caller. Keeping an ear on it.

[dibsmft]

I originally posted changes to 2 SIP variables (SIP usr ID for INVITE = No, Allow SIP messages from SIP proxy = No), but that alone didn't work. I still experienced this problem.

After reading some more posts, I made this change in Grandstream configuration:
(on page 4 - FXS PORT) NAT Traversal (STUN) = changed to NO
(on page 3 - Advanced) STUN server is = <blank>

I think some users recommend to use STUN, but my problem was that is was mixed up, NAT Traversal (STUN) was set originally to YES, but there was no server.

It is working now for 6th phone call that I received, I could hear the caller. Keeping an ear on it.

This is the area that is associated with creating/fixing the problems being observed (especially for the Linksys ATAs). The thing to do is experiment and fin out what works. The provider that I use has there own STUN server and I need it for the GC286 but not for the Android app. The Linksys 3102 needs STUN and NAT traversal etc. On the other hand my second provider does not use STUN and needs no special configuration.

That is the problem with setting up SIP voip .... so many different ATA devices and even more router models (some of which seem to try hard to be unfriendly to voip).

[TKS]

After a day of working well for all calls, i just got 2 calls from numbers that haven't called me before, and the problem is still there. They can hear me, but I can't hear them. This is not only annoying, but also very embarrassing, I have to tell the caller to hang up and I will call them back. It's been like this for over 2 weeks, can Fongo SUPPORT answer this type of question, at least point us in right direction for troubleshooting, because I try a new setting every other day or so, and many many other users are having the same issue !!!

Please someone help!

[iDrone]

Yes, TKS, same here, very frustrating!! Everytime I change a setting it seems it works, but it's only temporary or only for certain people. Very bizarre and quite maddening not hearing anything back from Fongo. Obviously something has changed on their servers, as my ATA device + Router were working just fine since I set them up, then after the recent Fongo problems/outage, it just hasn't been the same. Fongo's advised settings (http://forum.fongo.com/viewtopic.php?f=15&t=647) don't work either :-/

[iDrone]

I ended up putting my ATA into DMZ in my router, I've made and received several calls, so far so good. I think there is more port forwarding going on than the 5060-5061 and 13000-13001 ports I keep on reading here on the forum.

[Jake]

I ended up putting my ATA into DMZ in my router, I've made and received several calls, so far so good. I think there is more port forwarding going on than the 5060-5061 and 13000-13001 ports I keep on reading here on the forum.

I think the 13000 and 13001 ports are for the softphone. You really need to find what RTP ports your ATA uses and make sure those are forwarded in UDP. My PAP2T uses 16384 to 16482.

Changing something or rebooting will just re-sync the router and ATA. After a while it sounds like the router is forgetting what it should be doing for the ATA; which is why the right ports need to be forwarded. If you find that having it in DMZ has fixed things, then you need to find the correct ports and try that. Leaving it in DMZ isn't really the right thing to do.

[iDrone]

Thanks for your reply Jake, much appreciated. According to my ATA (Grandstream HT286) the RTP Port is 5004 (which is the default) I had forwarded this port as well previously, but unfortunately to no avail. However, just below it has the setting "Use random port:" set to "Yes", which kind of tells me that even though it is set to 5004, it can still use a random port in the 1024-65535 range!? I have read several things about the random port settings, some (mainly SIP phone companies) say you have to set this to "Yes", and some (mainly users) say to set this to "No". Since the settings Fongo advises, leaves it to "Yes", I kept it this way, but I can try changing that. I can also just forward the whole 1024-65535 to my ATA and see if that works, although I am sure my PS3 is using ports somewhere in that range as well, so probably not such a good idea.

Having the ATA in DMZ isn't really that bad in my opinion, the only risk is that someone could hack into my ATA, and change the settings. However, me using the ATA solely for personal use, I am really not that big of an interest for hackers. Also, even when hacked, it doesn't give access to the rest of my network, so really not that big of a deal. Nevertheless, I will try making it work without DMZ, but I had to do something in the meantime, as I was hardly reachable for anyone.

On a side note; I have been using FPL for nearly a year now and never had to do any port forwarding of any sort for the ATA and it was working perfectly fine, also after rebooting the ATA + Router + Modem. The port forwarding problems only started after the FPL outage we had, therefore I assume Fongo has done some changes to their server settings (maybe port changes) which made my Router suddenly block these ports.

[Jake]

I know what you are saying about having just the ATA in DMZ, but the way I have always looked at it is that if something else on my network needs certain ports, then having something in DMZ might mess those up. Obviously having a computer in DMZ is a different ball game, but I would not be surprised if someone out there has a way of using your ATA against you if enough access is granted.

I don't have a Grandstream to play with and help you, but I am betting that port forwarding is the area you should be looking in. Finding the right settings for things though may take some time, especially when you probably find that when you change something it works for a while before it decides not to again.

[iDrone]

Well, well, I am pretty sure I finally figured it all out. Since my phone started to fail even when my ATA was in DMZ and my PlayStation was acting weird as well, even with all the port forwarding in place, I decided to try to fix whatever was wrong with my PS3, with the hopes that would solve my phone problems as well. Finally I found somewhere on the PS3 forum to check the mode of the modem (Routed vs Bridged), because when in Routed mode, it will act as a router and conflicts the settings (like port forwarding) you do in your actual router. So, I changed my router to Bridged Mode (as it was indeed in Routed Mode), and changed my Router to handle the PPPoE connection. After the settings were saved, I first tested my PS3 and it gave NAT Type 2 (exactly what I needed ) and the phone registered right away as well and is so far processing inbound and outbound calls no problem. The fact that I didn't need to reboot everything, but things started working right away as soon as I had made the changes, gives me very good hope for my phone as well, as before I had to do a hard reboot on my Modem/Router/ATA after every change for things to work again, which was only temporary, because after a little while things started failing again. Anyway, thank you all for your help, and I hope my post may help others, that are still dealing with this.