DID to Voxalot

[rogerp]

Hello,

I have my account as VoxLite, and i have a DID number forwaded to my voxalot account though SIP.
It is forwaded ok, but when dialling the DID, no ring occurs although line is free, and voicemail message "this user with extension xxxx is unavailabe" sounds.. am i missing some configuration on the voxalot? how come incoming calls don't come throuhg?

Thank you!

[emoci]

Originally Posted by rogerp

Hello,

I have my account as VoxLite, and i have a DID number forwaded to my voxalot account though SIP.
It is forwaded ok, but when dialling the DID, no ring occurs although line is free, and voicemail message "this user with extension xxxx is unavailabe" sounds.. am i missing some configuration on the voxalot? how come incoming calls don't come throuhg?

Thank you!

You say your did is forwarded via SIP.

In that case it should be forwarded to
123456@voxalot.com or even better 123456@xx.voxalot.com (where xx is us, eu or au depending on which server you are using)

Of course replace 123456 with your VoXalot number.

Now, can you receive calls directly to VoXalot though? :
-Find a SipBroker Number local to you (see SIPBroker - PSTN Numbers )
-Call it
-When prompted enter *010-123456 (where 123456 is your 6 digit VoXalot number)
-Does your account ring this way?

If it still does not ring this way, it is likely a NAT issue, see here for some suggestions:
http://forum.voxalot.com/voxalot-sup...html#post14452

Depending on which ATA you are using take a look here as well: Hardware Configuration Settings - Voxalot FAQ

Hope this helps...keep us updated how it goes

[rogerp]

Hello,

Thanks for the reply!

In effect i tried to call the SIPbroker, and it still got redirected to the voicemail.

Everything was set as you said... i searched through the ATA link mentioned. I have a Siemens c470IP phone, through a router, somewhere i read to best set stun server, after i set the stun server all calls come through.

Problem solved!

Just one doubt, i am using stun.voxalot.com server, i guessed it is the one, is this the correct one? i am using eu.voxalot.com
It works now so i guess it is good.

Thank you!

[green]

Originally Posted by rogerp

Just one doubt, i am using stun.voxalot.com server, i guessed it is the one, is this the correct one? i am using eu.voxalot.com
It works now so i guess it is good.

Thank you!

You can use any STUN server. It's not tied to your provider or SIP at all.
Use one which is 'closer' to you.

[rogerp]

Hello,

Now I have incoming audio problems.
Incoming calls come through ok, i don't hear the caller but the caller hears me.
I have searched thouroughly the forum, and tried to continue this post without luck: http://forum.voxalot.com/voxalot-sup...ing-calls.html


I have tried the following:

-Try calling a SIPBroker number, than *010123456 , 123456 replaced with your VoXalot number. Do you receive this call, and is this also suffering from audio problems?

Yes, i receive the call, and i have 2 way audio, i hear the caller, the caller hears me.

-Try entering your voxalot URI 123456@voxalot.com in the link below. Instead of you calling Echo, echo will call you so you can see how it behaves when it is an incoming call: SIPBroker - EziDial

Yes i hear echo.

-If your router has UPnP I would suggest activating that rather than DMZ

Activated.

-Preferably make sure you have STUN set up (stun.xten.org has been working for me)

Already setup to stun.voxalot.com

-This is optional, but it may help to open these port ranges and forward them to your ATA's IP:

5050-5064
5000-5005
16300-16500 (this maybe slightly different for your ATA, there should be an RTP port range setting on your ATA, if it is different, note the range and open that range instead in your router...)
For some background on this see: http://forum.voxalot.com/12685-post20.html

I use a Siemes c470 ip phone, ports are 5060 and 5004, they are open in my router, also opened the stun port 3478, just in case.

-Make sure there are no Internet Connection problems. Run a SpeedTest, or maybe VoIP test at TestYourVoIP.com

no problems.

-If Nat Symmetric Hanlding is set to NO and Optimize audio to YES (these are both setitng s in your VoXalot acocunt), try setting Nat Symmetric Handling to YES and Optimize audio to NO

NAT symmetric is set to NO, i don't see option to optimize audio, where is that?

-Make sure that the codec string you have set up for Netelip matches what they support (personally I've had good results with this string: ulaw;alaw;g726;g729;ilbc;gsm, give it a try )

It seems so.

So if i receive a call from a landline to my DID, which is configured to go through sip voxalot, phone rings, i pick it up, but i don't hear nothing, although the caller can hear me.

Please help.

Thank you!

[emoci]

Who is the DID provider?

-Try using stun.xten.com instead of stun.voxalot.com
-The stuff about codecs, and Optimization applies to outgoing calls not incoming (so do not worry about it right now)
-On your ATA see if you can find these two fields and enable them (set them to yes)
NAT Mapping Enable
NAT Keep Alive Enable
-Try having NAT Symmetric set to both NO and YES and see if anything changes for you
-Is the DID being forwarded to 123456@voxalot.com or 123456@eu.voxalot.com?

[rogerp]

Who is the DID provider?

Did provider is DID World Wide (didww.com)

-Try using stun.xten.com instead of stun.voxalot.com

Yes tried that no luck.

-The stuff about codecs, and Optimization applies to outgoing calls not incoming (so do not worry about it right now)

Ok, thanks for clarifying.

-On your ATA see if you can find these two fields and enable them (set them to yes)
NAT Mapping Enable
NAT Keep Alive Enable

I don't have those fields, fields i do have when setting voip account in the phone software are:
- NAT refresh time (set to 20 seconds)... see image attached.

-Try having NAT Symmetric set to both NO and YES and see if anything changes for you

Yes i tried that before, nothing changes in either setting.

-Is the DID being forwarded to 123456@voxalot.com or 123456@eu.voxalot.com?

Did is forwaded to 123456@eu.voxalot.com

Is it normal that when i call through sipbroker it gets through but from a land line not? i thought maybe it was a router problem, but that makes me think not.

Thanks for your help.

[emoci]

Hi,

I noticed something...

First I was reading through
http://forum.voxalot.com/voxalot-sup...-received.html
http://forum.voxalot.com/voxalot-sup...-question.html

It seems DIDWW has a test page DID World Wide, International DIDs forwarded to PSTN and VoIP by SIP, IAX, H323, Skype, Gtalk, MSN Messenger. Origination services, SIP DIDs

So I ran a test with my VoXalot acct., and what I noticed was that calls to my VoXalot acct. were only successful when I chose USA as my preferred server

[rogerp]

Hi,

Yes i tried that but i could get that test to work.

I also contacted my DID provider, and they said it looked like a NAT problem in the router. As i already had opened the ports related 5004/5060, i don't know which other ports are used. So I accessed my router, and opened a wide range of ports for the phone ip, and I have finally got it to work!

Thing is, which port can it be using? how can i find that out, i don't like to have all ports open like that.

Thanks.

[emoci]

Originally Posted by rogerp

Hi,

Yes i tried that but i could get that test to work.

I also contacted my DID provider, and they said it looked like a NAT problem in the router. As i already had opened the ports related 5004/5060, i don't know which other ports are used. So I accessed my router, and opened a wide range of ports for the phone ip, and I have finally got it to work!

Thing is, which port can it be using? how can i find that out, i don't like to have all ports open like that.

Thanks.

These are the ports I have opened:

5050-5064 (eg. this is a range form 5050 to 5064)
5000-5005
16300-16500

Notice 5004/5060 are just registration ports, audio tends to go through the 16000 range ports in my PAP2...

Poke around your ATA and see if there are any settings for RTP ports, then simply open those in your router...

If you can't find anything that way, while on a call, open your ATA's interface, it should normally report what ports it is using for the RTP stream...then simply open a range aroung the port it is using

But I guess the good news is that you now know what the problem is (it's just a matter of limiting the port range so you are not leaving too big a hole opened up)