Need Help - random incoming calls do not ring through

[johnboykw]

So...not getting a warm fuzzy of late.... Had a PAP2T-na and had determined it was causing me sound issues.

Got a new HT702 by Grandstream - sounds great, but now I get random incoming callers complaining that they can't reach me, either it rings continuously, or it rings a few times, then gives a fast busy. I've tested this (with my cell phone) and can confirm this is what's happening.

All settings are set to the device sharing pdf provided by FPL, with the exception of NAT traversal is set to "No, but send keep alive", as I needed to set this to get ANY incoming calls to come in.

Can someone from FPL please tell me to PM them to get this resolved? WAF is dropping rapidly now and I need to get this fixed ASAP!....

Thanks,
John

[dibsmft]

It seems that the incoming audio packets are not making it through to the ATA. So, have you tried setting NAT Traversal to yes and what result did that give?

[johnboykw]

If I set the NAT traversal to yes, then it would seem that I need to specify a STUN server, of which I don't know what to put in there (as from what I understand, FPL doesn't support this, but someone please correct me if I'm wrong here).

[ddd]

ATAs must power charge to make bells ring: intermittent phone shorts?

It also happens with certain cordless phones.

ddd,
VoIP TS for other provider: many calls per day

[dibsmft]

If I set the NAT traversal to yes, then it would seem that I need to specify a STUN server, of which I don't know what to put in there (as from what I understand, FPL doesn't support this, but someone please correct me if I'm wrong here).

Any (almost) stun server will do eg. stun.ekiga.net or stun.callwithus.com . You may need to enable "NAT mapping enable" as well if you have it - OK I don't think the GS 702 has it but STUN should be all that is needed.

[johnboykw]

ATAs must power charge to make bells ring: intermittent phone shorts?

It also happens with certain cordless phones.

ddd,
VoIP TS for other provider: many calls per day

Thanks for the idea, but I can safely say this is not the issue as I had an different ATA in use before, where "this" problem didn't occur.

[johnboykw]

If I set the NAT traversal to yes, then it would seem that I need to specify a STUN server, of which I don't know what to put in there (as from what I understand, FPL doesn't support this, but someone please correct me if I'm wrong here).

Any (almost) stun server will do eg. stun.ekiga.net or stun.callwithus.com . You may need to enable "NAT mapping enable" as well if you have it - OK I don't think the GS 702 has it but STUN should be all that is needed.

I'll try the callwithus and see if that makes a difference. I've done some reading though and there seems to be mixed reviews on using STUN in the first place though. Anyone from FONGO willing to comment?

[dibsmft]

Stun usually can help with the sort of problem that you have but not always. Sometimes it can cause/create new problems especially if the provider or device already uses NAT traversal methods. It is always worth a try because it can easily be removed. While none of the three voip providers I have on my desk IP phone need STUN, the one of the three that I also have on my Grandstream 286 needs STUN to run there and works faultlessly. The only thing you can do is give it a try .

[johnboykw]

Well, had set that up and still hasn't fixed the issue I'm afraid. Any other ideas?

[dibsmft]

What does the 702 status page tell you?
Can you post the configuration page from the 702 for the FXS port to which FPL is connected showing you current settings and a list of the ports that you have forwarded from the router. Did you give the Grandstream a fixed IP from your router?

[johnboykw]

Here's the status page:

Product Model: HT702
Hardware Version: V1.4A Part Number -- 9610002014A
Software Version: Program -- 1.0.1.6 Bootloader -- 1.0.0.7 Core -- 1.0.1.4 Base -- 1.0.1.6
CPE --
System Up Time: 14:16:03 up 2:56
PPPoE Link Up: Disabled
NAT: Port Restricted Cone NAT
Port Status:
Port Hook Registration DND Forward Busy Forward Delayed Forward
FXS 1 On Hook Registered No
FXS 2 On Hook Registered No

I'll try to get a screen capture of the config of the FXS port on the next post.

Port forwards to be on next post too.

[johnboykw]

config1.JPG

config2.JPG

config3.JPG

Con't on next post

[johnboykw]

config4.JPG

portforward.JPG

K, that's all of it.

[johnboykw]

Notes: SIP3 is for my second line (non-FPL) and SIP4 and SIP5 was recommended by portforward.com, as per instructions from a Fongo Tech support agent I have already spoken to today.

[dibsmft]

Well, I will try some questions and suggestions,

Does the ATA actually get the 192.168.78.50 IP address when it logs in ... every time ie. is its address fixed for sure?
With multiple devices I don't forward port 5060 to anything to save confusion so I would remove 5060 in the SIP1 and change it to 5061-5064 UDP. In the ATA config the internal SIP port becomes 5061 for line 1, 5062 for line 2 etc. Give each line a different RTP port.
You have stun enabled, for now disable it. Did you specify a STUN server somewhere? You will probably need to add it again later.
Otherwise things should work with your present configuration.

[johnboykw]

192.168.78.50 is a static address for the ATA - so ya, it's fixed for sure.
Not sure what you mean by "multiple devices" though... it's just one ATA. Also not sure what the purpose of not allowing 5060 through and changing this to 5061? Isn't the port just a number? Each line does have different RTP ports already.

And alright, STUN is back to "no, but send keep-alive". Ya, forgot to mention the STUN server is specified in the advanced page. Let me know if you want to see that as well, as that's the only thing I changed there.

[dibsmft]

The device has two lines and I was simply mentioning that multiple lines do work OK and independently and you have only two lines. I don not own an HT702 and my HT286, though similar is not quite the same and has only one line. On a provider somewhat similar to FPL (FPL service is very thin in NL) my GS286 is set with SIP 5061 UDP and RTP 8006 and it is running STUN.
So, probably best to go back to square one and set things up in order.
1. Power down the router and the ATA start the router and check that the ATA gets the 192.168.78.50 address. If not find out why not and correct the problem.
2. Clear the port forwards in the router and change them to
Initial minimal setting (the settings that I would start with):
5061 - 5062 UDP to 192.168.78.50
8000 - 8020 UDP to 192.168.78.50

Set line 1 on the ATA to SIP port 5061 and line 2 to SIP port 5062 and RTP ports to line 1 to 8006 UDP and line 2 to 8016 UDP. Use random RTP port "No"

Most SIP services should be able to work with those settings. Turn the ATA and router off and the reboot first router and then the ATA.
See if the ATA registers and works.
If it fails then set STUN with the stun.callwithus.com server to run on the ATA, and then turn the ATA and router off and the reboot first router and then the ATA. Try the ATA again.

If these basic settings don't work then we can consider other things. You have forwarded lots of ports but they should not be needed, however, perhaps they are with FPL.

[Bloodsong]

I can confirm the only necessary ports for the ATA to work are your SIP ports as set in your device configuration for the individual lines. (Usually, but not always in the 5060-5080 range) and a UDP range used for the RTP-stream. (This is what actually carries your voice. Also generally defined in your ATA, Linksys often places it in the 16000+ range somewhere, common practice is 10000-20000, which is WAY more than needed, but works great if you're running your own IP-PBX I see yours is set to 5004 and forwarded from the router as such, 5004-5005 are designated for RTP by IANA thus the Linksys choice here.)

the 13000-13001 ports are specific to the FPL branded Softphones.

Wow that is an incredibly short re-registration period.
Completely un-related, you don't like MWI?

I see you're on a dd-wrt router.
Turn on syslogd under services and under security/firewall turn on the log, I'd hazard a guess at "medium" logging level, for dropped and rejected packets.

Then call yourself until you have the fast-busy issue (or continuous ring) and check the logs. (I don't know if the logs have GUI access, you may need to SSH/telnet into the router, logs are stored at /tmp/var/log/messages

BTW: I'm at work right now so can't check, did you remove the EXIF data from those JPEGs? If not I can recover your SIP Username *evil grin*