incoming calls to my voxalot number without sound
 

Hi, my incoming calls to my voxalot number 315409 dont have sound (outgoing and incoming voice), i try call me from my other account in voxalot (315407), but is the same, not sound, the call cut to 10 seconds. But if i call with sipbroker format (*010315409), the call is connect and have sound in both leghs. What i can do? Thank you for your reply

PD
In this account (315409) i can make outgoing calls, and work ok, my line have sound in both leghs (incoming and outgoing)

resolute problem. Thank you

Hi

Same problem here, no sound on incoming calls on my account 924066 ! - Yesterday it works fine with sound both ways. But today there is no sound in/out on incoming calls !

Outgoing calls work fine with sound both ways ! And i have not done any changes here...

Any ideas ?

Hi wycodreams

What was the solution to get your incoming call/sound to work again ?


As i write above i have exact the same problem, the phone rings normal, i answer, but there is no sound, and the connection broke after 10 seconds.

Outgoing calls works without problems. - And i'm very shure that the problem is on Voxalot's system because it works fine the days before, the problem started yesterday. - And i have 2 other Voip providers on the same router and phone (Grandstream GXP-2000) they both works fine, with sound in/out on incoming and outgoing calls...

I have fore the moment changed to another provider until Voxalot has fixed the problem.

Hi, i register my voxalot account in my ATA and mysipswitch. This is the problem. I delete my settings in mysipswitch and the problem is finish. Bye

Do you solve your problem? I have the same problem, yesterday it works fine. but today I can't receive the phone call. while no problem to call out. what's going on?:confused:

Hi

No ! - I still use anohter provider.

I'm shure that i cant' do anything about it, the problem must be related to Voxalot's system.

Ole,

Where is the inbound call coming from?
.

my dids are from DIDWW.since yeasterday, they are all out of function. Ring normal, but pick up the phone no sound. another side is also no sound. but I can call out with no problem. Martin, Do you know what is going on?

Have you checked if the DIDs are working correctly?

Yesterday, I needed a DID for testing and I used a DIDWW test number using the Midle East server and it was not working.
The test using the US server worked fine.
Just a thought ......

I am using US server. I didn't change anything. The problem is starting from yesterday 1:30pm in montreal.

Hi Martin

It comes from: Musimi.dk

hi, oleras. I fixed the problem. I think it is because the stun server changed. now you have to use stun.voxalot.com au:3478. For my case, I changed to it, it right away work perfectly. But I don't understand I use my original one for 3months. there is no any problem. it was just starting yesterday the old one doesn't work for voxalot system. Maybe it is because voxalot upgraded the system.
good luck

old one? not sure what you mean by this?

The IP address changed 2 weeks ago, is this what you are referring to?
.

No, martin. I mean the old stun server which I used before. It used to work perfectly, but just a few days ago, suddenly I can't receive the phone call. But after I changed the stun server into stun.voxalot.com, the income call came back.

Hi

Ok, normally it's possible to use any stun server, i use one here in Denmark, but will test stun.voxalot.com.

Well, my problem is that things work perfectly, with sound both ways, f.ex. 3-4 days, and then the 5 day there is no sound on incoming calls, next day again it work ok aso. It's not stable !.

/Ole

Calls from did worldwide not working
 

I noticed that incoming calls from did worldwide ring but not sound then drop after a few seconds or go to voice mail.
Outgoing calls work fine.
Have been working fine in the past - no changes in set up of my sipura 2000.
Does anyone else notice this.

please read no.13

Incoming from worldwide did nos sound
 

:(
Thanks - I changed the stun server to "stun.voxalot.com" but get the same negative result.
The phone rings, I pick up and hear some noice but then it hangs up ...does not go to voicemail anymore.
I contacted wwdid and they claim "they tested and the phone rings" and they do not provide further help.
I am using a Sipura 2000 and can make outgoing calls fine (voxalot>voipdiscount.com)
As said it had worked for a few months before stopping.
I could furnish SP2000 config if necessary. Cany anyone assist?

For those of you having problems, how do you have your home router setup for your VoIP?

I'm not sure if it's the problem here. However, it's been my experience that 1-way (and 0-way) audio problems are OFTEN the fault of your home router, that is incorrectly treating the audio stream as an "attack" from the internet, vs properly forwarding it to your VoIP equipment. i.e. in many cases the audio is actually getting to you, but your router (or the router of the person on the other end) is preventing the audio from making it the whole trip between the two VoIP end points.

Now, if the above is what is going on, than there are many settings that could be done to the various VoIP equipment, and probably even a few changes that could be done to VoXaLot, to help convince the routers to "behave". However, such changes are not certain to work, and would really only be "work arounds" away. IMHO the proper "fix" for this (because it's the only "fix" that addresses this issue 100% of the time, because it works directly on problem routers themselves), is to have people configure their routers to always allow this VoIP traffic (and always send it on to their VoIP adapters). And that is usually done with "port forwarding" on your home/NAT router.

Specifically, VoIP is naturally more reliable if/when SIP and RTP ports of the VoIP adapters are exposed to the internet (although you generally do NOT want the web interface of VoIP adapters to be exposed, just the VoIP ports, which is why I usually avoid using the router's DMZ and instead "port forward" the SIP and RTP ports)! The way port forwarding is done varies with your router, and which ports to use varies with which VoIP adapters you have, and how you have them setup. So I can only give general advice on this. However, in general, it should be possible to modify this technique for other adapters and routers, the details will just vary slightly.

1) In general you need to first assign a fixed IP address on your LAN to your VoIP adapter. This is often necessary, so you have a consistent place to tell your router to forward VoIP ports to. Because if you let the router auto-configure based upon DHCP, it might get a different IP address (on your LAN) when you reboot the router, and therefore mess up your attempts to "port forward" (because the adapter has "moved" on your LAN, as it were). This is usually done on your VoIP adapter's "advanced" settings, and you should try to pick a LAN IP address that is part of your LAN (which with a lot of routers means that the 1st three numbers are the same as other devices on your LAN), but which isn't used for any other device on your LAN, and also is not part of the range that your DHCP server (that is often part of your router) auto-assigns to devices (because you will get real network headaches if/when two devices get the same IP address on your LAN).

2) Once you have the VoIP adapter at a fixed LAN address, you can proceed to "forward" the SIP and RTP ports for your adapter, to the IP address (on your LAN) you now have the adapter on. If you already know the UDP ports used (by your VoIP adapter) for SIP and RTP, than you should be good to go. If not, you will have to do some investigation in your VoIP adapter, to find the info.

2a) The SIP port is the port used to setup a call (i.e. ring your phone, and similar issues). It is often UDP 5060, but many adapters will let you set this to other numbers. And even with the "default" settings, many adapters use a different port number for the 2nd/3rd/etc line, than they do for the 1st VoIP line. So it's usually easiest (and most accurate) to get this number (or these port numbers, if your VoIP adapter supports more than one line), by just checking the SIP port settings in your adapter (with many adapters, this info can be seen in the advanced settings of the adapter's web interface).

2b) The RTP port RANGE is the range of UDP ports that can be used for VOICE TRAFFIC. With most adapters, this range is (by default) MUCH bigger than it needs to be. For example, some LinkSys adapters is 16384-16486 (over 100 different ports) by default, and some devices even use 10000-20000 (about 10,000 different ports) by default. IMHO even having 5 times as many ports as you have "lines" in your equipment is over-kill, so you really do NOT need anywhere near as many port options as VoIP adapters like to use "by default". Instead, you can often considerably lower this range (if desired), to give you less ports to "forward" to your adapter. And you can also edit the range, to allow multiple VoIP adapters to exist on your LAN, but with different RTP ports (so you can have your router forward the proper voice traffic to each adapter). I actually did the latter, as I have two VoIP adapters on my LAN (which I have setup with different, non-overlapping RTP port ranges, each properly "port forwarded" to the proper VoIP adapter on my LAN).

NOTE: On many LinkSys adapters, the RTP port range is defined/set on the "SIP tab" (of the web interface of the adapter), as being between the values "RTP port min" and "RTP port max". On other adapters, you will have to consult the adapter's manual to find the RTP port range.

Again, most of the above is often "not necessary", as many routers will often open the required ports dynamically as needed (based upon the actions of the VoIP adapters). However, it is the approach that seems to give the most consistent VoIP results, and therefore the approach I take on my LAN. Because without the above "port forwarding", you are counting on things like the "keep alive" packets from your adapter to keep the call channels open. And that can be "hit or miss" as to if it will work with any given router. But when you explicitly allow all VoIP call traffic to your LAN to be sent to your VoIP adapter (setup technique above), than issues such as "missed calls" (your router blocked the SIP port was blocked when it tried to "ring your phone"), or "1-way audio" (your router blocked the inbound voice from the other end) often go away.

NOTE: And as long as you only open the SIP and RTP ports (and then only to your VoIP adapter), you really aren't lowering your internet security much. But such an approach will give inbound VoIP traffic (which includes receiving the voice from the other end of the "call") a more open (unrestricted) path to your VoIP adapter.

Of course, this isn't a "cure all", and sometimes even this won't be enough to get fully working voice. But in many cases, the router on your LAN is the culprit that is causing the problems (by blocking the inbound VoIP packets needed for the call). And often the better the router's "firewall", the more likely the router is to cause problems in this respect (so higher end routers can actually be WORSE in this regard). But if you explicitly "port forward" the VoIP traffic (to your VoIP adapter), the problem often goes away. Because you have then told the router to always accept that traffic (and always send it to your VoIP adapter), even if/when the router thinks it came in "unsolicited" from the internet.

Problem using DIDWW with VOXALOT
 

Same thing happens to me.

1. If I phone from a computer or using sip broker, the incoming call goes through OK.
2. if I phone from some cellular phones it goes through Ok.
3. BUT IF I PHONE FROM A PSTN LINE, IT RINGS 1 TIME AND THEN IT FALLS DOWN.

Have someone found a solution for this?

So, I guess you guys don't recommend DIDWW...?

problem with voxalot and didww
 

I dont know if the problem is DIDWW or if its voxalot.