2talk Issue: Not receiving 95% of incoming phone calls

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Leese

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#81049 7-Apr-2011 09:41

Hi all,

Just wondering if anyone can shed some light on this one for me? (I don't seem to be getting anywhere with 2talk Support).

Our problem is that we don't recieve all inbound calls. 95% of our inbound calls go straight to the voicemail, so our telephone does not even ring.

We can (99% of the time) make outbound phone calls.

The setup I'm using is:

Router: EnGenius IEEE 802.11g ECB1221R

SIP Box: Grandstream Handytone-496

I have changed the SIP Settings and Proxy and Registration details, as below (as advised by 2talk Support), but this doesn't appear to have helped at all.

SIP Settings

   * SIP Port: 50600

Proxy and Registration
    * Proxy:  2talk.co.nz:50600
    * Outbound Proxy:  2talk.co.nz:50600

Our 2talk number is always showing in the 2talk Live area as "online", even when someone tries to phone us but the call goes straight to voicemail.

Any help/assistance greatly appreciated :-)

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#456254 7-Apr-2011 09:54

It will be your router / firewall blocking incoming SIP connections, it works outbound all the time becuase the router allows all outbound connections, you will find that incoming calls will work as soon as you have made an outbound call as it now opened up the connection but will stop working after a few minutesbecuase your router will close down the pin hole for that connection after a while, use a keep alive on your setup and set it to 30 seconds this will keep the pin hole open, it will use a sip notify or options

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sbiddle

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#456345 7-Apr-2011 12:30

Try changing your registration time to 60 seconds.

As Phil explained your firewall is blocking the inbound traffic as the NAT firewall pinhole isn't staying open. Setting up a port forward to your ATA will also fix the problem but unless you can restrict inbound SIP traffic to only that from 2talk (and I don't think Handtone's can do this as they're a very basic unit) you run the risk of your phone ringing randomly as all the bots port scanning for insecure SIP connections try your device.

Leese

5 posts

Wannabe Geek

#456373 7-Apr-2011 13:15

Thanks for your help guys! Much appreciated.

I'm a real noob when it comes to routers/firewalls etc, can anyone tell me where abouts I go when I log into my router interface to change the settings you've described etc?

maverick

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#456474 7-Apr-2011 16:47

You need to log onto the grandstream box, you must have done this before to set it up initially, when you do on that page is 2 settings, Nat settings and Keep alive Interval , check those settings

Make sure That Nat Traversal is turned on as this si the important one, the timer should be okay at 20 sec's

NAT Traversal This setting decides whether the NAT traversal mechanism is activated. It
should be set to ?Yes? if the device is behind a NAT router. If no outbound
proxy is configured, a STUN server needs to be set to activate STUN detection
mechanism. Usually ITSP will provide these settings.
If this field is set to ?Yes?, then the device will periodically (every Keep-alive
interval) send a dummy UDP packet to the SIP server to pinhole the NAT.

Keep-alive interval Default is 20 seconds. The minimum value allowed is 10 seconds. This is the
interval of sending dummy UDP packet to keep the NAT ?pin hole? open.

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Leese

5 posts

Wannabe Geek

#456512 7-Apr-2011 19:07

Thanks Maveric - much appreciated!

I've just logged into my Grandstream and these are the settings currently in it. What should I be changing?

http://img851.imageshack.us/i/natdetails.jpg

http://img130.imageshack.us/i/natjl.jpg

maverick

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#456516 7-Apr-2011 19:21

Is nat set to yes?, if not it should be as this will send a keep alive to their proxy and keep the pin hole open

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maverick

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#456519 7-Apr-2011 19:27

Opps sorry missed the images, do you have a stun setting in the box as you may need to populate that with 2talks stun server, was never a big fan of the grand stream boxes :), if that dosnt work set the registration timer to 60 seconds, not a good way to do it but am surprised that the keep alive will not keep the connection open.

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Leese

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Wannabe Geek

#456545 7-Apr-2011 20:42

Stun Server is set to: "stun.2talk.co.nz:3478" and the "keep alive interval" is set to 20 seconds.

One of the NAT settings shows as: "detected NAT type is symmetric NAT"

Once again, thank you very much Maveric! I really appreciate your help

maverick

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#456615 8-Apr-2011 05:58

Try removing the stun setting and see how you go, you may need to set up a static IP later bur see what happens with this setup first

If the NAT Traversal field is set to "Yes" with no specified STUN server, the HT496 will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the ?hole? on the NAT open.

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maverick

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#456617 8-Apr-2011 06:26

sbiddle: Try changing your registration time to 60 seconds.

As Phil explained your firewall is blocking the inbound traffic as the NAT firewall pinhole isn't staying open. Setting up a port forward to your ATA will also fix the problem but unless you can restrict inbound SIP traffic to only that from 2talk (and I don't think Handtone's can do this as they're a very basic unit) you run the risk of your phone ringing randomly as all the bots port scanning for insecure SIP connections try your device.

Yep they do have it Steve must have added it in afirmeware upadte, I always hated these boxes when we worked with them in the early days... cheap and nasty

Allow incoming SIP messages from SIP proxy only

Default is set to “No”. If set to “YES”, each incoming call will be checked first if it arrived

from authorized proxy. Useful for security installation to prevent incoming calls from not

authorized sources.

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nate

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#456719 8-Apr-2011 12:37

I don't think there are many techs out there that would give such comprehensive help to someone using a competitors platform, without giving them the sales pitch.

Great job, as always, maverick.

maverick

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#456724 8-Apr-2011 12:48

nate: I don't think there are many techs out there that would give such comprehensive help to someone using a competitors platform, without giving them the sales pitch.

Great job, as always, maverick.

Someone has to help them , and I don't do Sales (evil place), well not here anyway

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webwat

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#457145 9-Apr-2011 23:43

When I setup mine i had to turn off STUN and NAT on my voip phone's settings, everything started working after that and the registration kept the port open.

Time to find a new industry!

maverick

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#457156 10-Apr-2011 06:34

That would be a different issue webwat as it's unrelated to SIP pinholing, most likly the nat / stun setup where not working correctly with the router so the port info was not correct and not passing the SIP to the correct ports, also probaly nat was not needed as most routers have some sort of SIP ALG, these have been pretty poor in a lot of routers but are a lot better in the last few years.

This issue looks to be purely pinhole related as the device is working correctlybut shuts down for inbound after a time until an outbound SIP packet is detected, timer starts again generally these will shut down at about 60 sec or so depending on the router and I myself have never heard of a "EnGenius" but this will be my educated guess

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sbiddle

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#457166 10-Apr-2011 09:25

Just be aware as well that from past experience I can tell you that ECB1221R's make terrible routers. They do have a habit of locking up if there is excessive traffic put through them, particularly UDP (which covers VoIP).

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