To Voice mail, no ring

[Yroc]

Calls have been going to voice mail randomly (some do, some don't). When they do, I find that resetting the ATA fixes the problem at least temporarily. I have a static IP and ports forwarded. I should add that the phone will ring for the caller 4 times before it flips to VM, it is simply that my phone does not ring. Has been an ongoing problem for some time now.

ATA is a PAP2T
Router is Linksys wireless N-Gigabit WRT310N
ISP is 3Web through coax (ie. rogers).

Any suggestions?

[TheHardy]

Off the top of my head, the only thing that I can think of is that the ATA is losing registration to the network ... check your keep alive times on your settings.

Short of being near your ATA and seeing if it goes into de-registered/un-registered state, and then making a test call to confirm this .... but your statement about resetting the ATA seems to back this up, as resetting it forces it to re-register ...

So whether it is the keepalive, or something else, it seems to me that your ATA is somehow losing it's registration to the network ...

[Yroc]

One number that calls which never rings is from #70524240xx

The # is from the Canadian Magicjack.

[TheHardy]

One number that calls which never rings is from #70524240xx

The # is from the Canadian Magicjack.

Iristel-based number ... is it usually calls from the same #'s that hit voicemail without ringing, or is it more like _all_ calls?

If it is only calls coming in from specific numbers, then I wonder if there is something to it or not ... hmmm.

Any clue from any of the log files or anything, as to whether the state of the ATA is changing?

[dibsmft]

My number is of the Iristel stable as far as I know and it works perfectly on my Yealink desk phone but I did have problems with it initially when trying to use the number on a Linksys SPA3102 while it was OK on a Grandstream HT286. However, it might be that the caller is using a voip based system and there may be a configuration/router problem at both ends. One thing to try as the ATA is Sipura/Linksys based is to run STUN (stun.callwithus.com or stun.ekiga.net), "Nat Mapping Enable" and "NAT Keep Alive Enable" and see if this helps.

[Jake]

It could be you are having codec problems. Make sure you don't have you ata set to only use one preferred codec. I had this problem when I first had fpl because their setup instructions said that is how it was supposed to be. Some calls from VoIP users were going straight to voicemail.

[dibsmft]

Jake is correct but that was not the problem that I had with my provider unless it was at their end. The "Auto" setting on the ATA should be good enough with perhaps G711U as the first choice. "Use preferred codec only" should be set to the default which is "NO".

[Yroc]

Everything seemed ok for a while, but today I had another case of no ring (direct to voice mail). This time from caller 613-225-7xxx. I read the info above and tried changing

NAT Mapping Enable from no to yes and
NAT Keep Alive Enable from no to yes

I reset ATA and router, will see if it changes anything.

Also note that when I try stun.callwithus.com I get "It works".

My ATA was not set to only use one preferred codec

[dibsmft]

What you have done should have cured the problem. If not I am not sure what to suggest ,

[Yroc]

Phone did not ring again today

[dibsmft]

My incoming number also originates from Iristel and it works OK. The setting that I have are now are
NAT traversal disabled (It used to need this but no longer does) you can try diabling it but I think the PAP2 may need it.
UDP keep alive enabled - and set to 30 sec
Subscribe period 800 sec
Refresher UAC
Login expire 1800 sec

Not quite the same as the PAP2 but the important one seems to be the UDP keep alive. Try setting the NAT keep alive Interval 15

I think your line is "going to sleep and you neep to keep it awake!

[Yroc]

Cannot find a "UDP keep alive setting", only a NAT one.

[Yroc]

Seems random.. Working today..

I have to wonder if the issue is on the other end.

[dibsmft]

Could be at the other end. Is the caller using a sip phone? The NAT keep alive on the PAP2 needs to be on.

[Yroc]

In one case the caller is on a SIP phone. In others, they are not.