All calls being sent to Voicemail
 

Having received several instances of people not being able to call me (straight to Voxmail ), I have called in to all my DID's with Providers registered in Voxalot and indeed all calls are going straight to Voicemail.

I have turned off Voicemail and I get nothing but silence.

I have removed all Call Forwarding, all calls to Voicemail.

I set Call Forwarding for all calls to my Voxalot account, all calls to Voicemail.

Can someone please explain what is going on. I have made no changes to my Voxalot or ATA settings. The ATA is on register to Voxalot

Best regards

Grahame

I have just called my ATA on it's second port ( mysipswitch ) and everything works fine.

Voxalot issue ?

Best regards

Grahame

Can you check the status page on your ATA to see if both accounts are registering correctly?

Are you using the same SIP/RTP ports for both accounts? Suggest to use different ports (Eg 5060, 5062 / 5004, 5006)
You may be getting a conflict because of different expiration times between the two services.

Perhaps you can also try a different Voxalot server, or shut off the other account for debugging purposes.

Hi Kurun,

Thanks for your reply.

Both accounts are registered properly in the ATA.

Both accounts use different ports: 5060,5062

Expiry times are both set to 3600

I have tried running Voxlaot with the other account turned off, no difference.

I have moved all my principal DID providers to Mysipswitch ( Account 1 ) and all works fine. I have had to do this for testing but more importantly I need to receive calls. Voip supports my business and Voxlaot is my principal platform for this. I need these services running.

*600 and *500 work fine, it seems the Voxlaot server is not passing inbound calls to my ATA ( thinks it's busy, off line ?? ).

Could Martin or someone please have look at this from the Voxlaot end. There are several instances in my account i.e calls direct to Voicemail, which illustrate the problem.

Best regards

Grahame

It still sounds like it might be a user-side problem. This same scenario happens for instance when your WAN side IP address changes prior to the registration expiry. Since your registration lasts 60 minutes, that means if your IP address tends to change more often than that, your incoming calls will all be in limbo, and go to VM or silence or cause a busy signal. I am not sure why your other provider would work while Voxalot didn't though, if you have the exact same settings across the board for both lines.

Try changing the expiration of the registration to 300 or 600 and see if that helps you.

Hi ctaylor,

Thanks for your response.

There is certainly something weird going on. I took Provider A off register with Voxalot, registered A with Mysipswitch and asked A to call me. A reported the call went through correctly, but all they heard was music and the call then went off into the ether. Nothing my end.

I then took A off register with Mysipswitch, put A on register with Voxalot and asked A to make the call again. A said they heard music, the call went through, my phone rang but with one way audio i.e they could not hear me.

Provider A says they are sending the call correctly and the call is being received correctly. They do not provide music, they feel the music is being generated by an Asterisk source. Clearly they will not comment on what Mysipswitch or Voxalot do with the call once they they have handed it over.

All comments gratefully received.

I'm going to strip the whole system down tonight, close down Mysipswitch and put all back together in Voxalot with a single registration at 300 expiry.

Best regards

Grahame

My Voxalot registered ATA is set for SIP registration expiration of 120 seconds.
I have had issues with losing registration when setting for longer intervals, even though my IP address is stable.

One way audio usually indicates that there is an NAT issue.
Since the phone is ringing for incoming calls, the Registration server is obviously able to find yor ATA, but the RTP data is not transferring correctly.

What type of internet service do you have?
Are you using a router before the ATA?
If you are using a router, it would be advisable to use a STUN server also.
Are you using one VoIP device only or multiple VoIP devices on the same network?

I have seen situations where a correctly set-up ATA will not work properly when connected to a DSL modem directly, but will work perfectly with the same provider if connected behind a router.
I have not been able to understand why, and it is my suspicion that some providers only allow specific devices to connect to their network.

Heh.

Hard-reset the ATA, update the firmware if a newer version is available, and reprogram it from scratch. Then figure out what is still acting problematically.

Hi guys,

Thanks again for your responses and my apologies for the delay in replying.

I now have a working system with Voxalot passing calls from my DID's with no problems. This has meant turning off one account in the ATA and removing one PC from my network. I am pretty sure the problem is either with the modem/router provided by my ISP or the ATA box itself.

The modem/router is a Livebox provided by Orange/France Telecom. This has a free VoIP service built into it, but Orange will not tell me settings or the technology being used. With two accounts active in the ATA and the free VoIP service running ( can't turn it off ! ), nothing works properly.

The ATA has an Ethernet pass through connection which avoids running cables i.e PC to ATA, ATA to Livebox ( it only has one Ethernet port ! ). With a PC connected to the pass through, irrespective of how many SIP accounts running, again nothing works as it should.

This is not the first problem I have had with Orange so come end of June I will be changing ISP ( and buying my own modem/router ! ). So unless anyone else out there is struggling with Orange in France, let's close this down.

I would however like your views on the ATA settings which are as follows:

Different SIP Local Ports for each account

Provider registration details as required

Expiration duration: 300
Register Re-send Timer:180
Session Expires:180
Min SE: 30

RTP Range: 3000 - 65535 ( This interests me as it's the same on both accounts, I can only set a range not a value, per account )

Nat Keep Alives:120

Stun configured on both accounts ( Voxalot and Xten )

Any thoughts on the above settings would be greatly appreciated.

Best regards

Grahame

My guess for my equivalent since your terminology is different (mainly default settings [Sipura]):
Expiration duration: 600
Register Re-send Timer:30
Session Expires:7200
Min SE: 1

RTP Range: 16384-16482

Nat Keep Alives:28

Hi ctyler,

OK Thanks for that, I'll play around with the settings.

I'm still interested in the RTP settings for each account in the ATA. i.e should the RTP range be a different for each account ?

Best regards

Grahame

Yes, both RTP and SIP should be different in each ATA account.
If every IP user agent (soft-phone, IP Phone, or ATA) is forced to use different listening SIP Ports (Eg. 5060, 5062, 5064, etc) and separate RTP ports (Eg. 5004. 5006, 5008, etc), the overall SIP setup will be more stable.

What I find most stable when multiple devices are being used:
1) Fix the private IP address of each SIP device
2) Set up SIP ports, RTP ports as indicated above
3) Use a STUN server
4) Forward the respective SIP and RTP ports to the individual device fixed IP address

Of course to do this, you must have full access to your router and private network.

I personally would avoid that Kurun, if I were you. It is an over-complicated setup that in fact is unnecessary.

You could have 10 SIP devices (ATAs, IP phones, and softphones) on your LAN at 10 different IP addresses, and each and every one of them could be using ports 5060 and 16384-86, and the router would sort it out just fine. All the devices (or computers) could get their local IP addresses assigned by DHCP to boot and it would still work itself out. That's the power of STUN and NAT routing. What happens is that to the outside WAN world 192.168.1.3:5060 would become something like 212.54.78.201:60445 and so even if you had a second device that was at 192.168.1.4:5060 it would be assigned something like 212.54.78.201:61225. In other words, there is more than enough port 5060 to go around when using a NAT/PAT router since all the ports get changed anyway to something else.

Network address translation - Wikipedia, the free encyclopedia
Port address translation - Wikipedia, the free encyclopedia

Hi guys,

Thanks again for your replies. I think you are both right, it seems it's a question of two different strategies to address the same issue.

I also found this in another thread from Martin:
Quote:
So just to re-iterate, if you want to properly receive the ACK message from the other end point, and you are behind a firewall, you must either:

1. Use a protocol like STUN to perform client side NAT handling (Note: STUN is broken in SJPhone and needs to be disabled. Make sure the "Use discovered addresses in SIP" is unchecked. I suspect this is half your problem)
2. Open up your firewall ports and forward to your device
3. Register your device with a proxy that has built-in NAT handling capabilities. End Quote:

I'm using STUN, have the ATA in the DMZ and have Voxlalot doing the NAT handling, so I should have all bases covered ( unless I'm overcooking it ! )

I've got my other PC now working by bypassing the ATA and VoIP traffic stable by just using Voxlalot. I'm changing ISP and systems end June so I'm going to leave it there for the moment. I need the system running and can't give any more time to testing at the moment.

Thanks for all your help with this, I'll let you know what happens after my system change.

Best regards

Grahame






.

Thanks Grahame. I am a firm believer in not touching your router's port-forwarding or DMZ settings when using VOIP, and letting STUN handle it all for you. The only occasion you should ever manipulate your router's port-forwarding settings are the rare cases when you are hosting a web or email server at your IP address and you need port 80 to get directed to the right machine. That means 99% of people don't need to manipulate those options in their router. These days, even bit-torrent or gnutella clients will manage port-forwarding for you (via UPnP), and with SIP and VOIP using correct STUN and NAT settings in your device will take care of it. Its also simpler with less to troubleshoot in case something goes wrong. My two cents.

I also came to that conclusion eventually. I used to have intermittent problems with incoming calls on my PAP2 until Martin advised me to disable port forwarding and enable STUN. This has been working flawlessly for many months.

Some clarification on my earlier post.
I always opt for the simpler solutions wherever I can, and I fully endorse CTylor's comments regarding letting STUN do the router sorting out.
My suggestion was primarily to address the problems I experienced when the same Voxalot account was logged onto by 2 or more ATAs/SIP phones operating behind the same public IP address.
For all I know, the problems may have been due to the particular router and SIP devices being used.

For a single ATA, I have generally found that using just DHCP & STUN is sufficient, without the need of any port forwarding.

For multiple devices logging onto different SIP services (Eg. Voxalot and FWD), STUN is also usually sufficient.

For whatever it is worth, Wikipedia states that "STUN will not work with Symmetric NAT".
Simple traversal of UDP over NATs - Wikipedia, the free encyclopedia

A word of caution regarding uPnP - Earlier this year, there were reports about a security flaw when using uPnP, and I have not found any indication that this is resolved.
Severe UPnP Flaw Allows Router Hijacking -- Computer Security -- InformationWeek
Flash UPnP Attack FAQ | GNUCITIZEN