Just today we have now opened port 80.

The SIP proxy ports now supported on the Voxalot side are:

80, 2060, 3060, 4060 and 5060 and 443 (as per Craig's post below)

Will check when traveling the next time, but sure will be a great help in Hotels and public Wi-Fis. Thanks for listening to the users. richard99 will do a jig... :-D
- gambrinus

Ha ha ha - I thought the same thing too :)

Thanks Martin, it is working tested in place where I was never able to use voxalot. ...still pending port ...:)

No problem. I'm curious why you would need 443 when you have port 80?

It is fine. As far as port 80 is serving its purpose. You are right there is no need for port 443.

I was just wondering, but isn't SIP based on UDP?

Serving SIP on port 80 would mean that it would be served on UDP port 80, and HTTP traffic is served on TCP port 80.

I don't think this will make much difference for ISP's only allowing port 80, as they will usually only allow traffic on the TCP port 80.

True but chances are, most ISPs are unlikely to touch port 80 at all just to block SIP.

Also SIP traffic can be either UDP or TCP. If your client chooses to use TCP as the transport, then the traffic will be SIP over TCP using port 80.

I am pleased to announce that I have just enabled forwarding of port 443 on all of our proxy servers. Please give it a try and let me know if you have any problems.

--
Craig

x-lite, call disconnect after 20 seconds when using other port 80
 

i am an area where default port 5060 is blocked so i have used the outbound proxy and port 80 for registration with voxalot it registers properly but after 20 seconds of call it disconnects with x-lite i and also i am not able to receieve incoming call. if any one can help i will be thankful

same... :(

in outgoing calls.

x-lite
 

:(hi
i can't receive any call on x-lite at pc.
but i can receive call on fring at my mobile.
pleas help me.
i change port x-lite too.
but i can't receive call too.
pleas help.

outgoing calls get disconnected in 21 seconds...
 

Hi,

Voxalot is a great service. Recently i came across a strange problem. All my outgoing calls get disconnected when going through voxalot, no matter which is the destination SIP provider.

I am using a Fritz. 7140 ATA and this is a simple setup. When i am able to use port 5060 for SIP everything works as expected (I've tested this thoroughly).

However, my internet / telephony provider is using port 5060 so it's practically blocked for my SIP providers. I am using Voxalot as a Gateway on different port (2060) and registers fine. Incoming calls get connected well. Outgoing calls work, Voxalot Dialing plan works, but no matter to which provider i get connected through the call drops in exactly 21 seconds...

Any suggestions are highly appreciated.
Best Regards,
~thanasis

Try using 80 or 443 and see if anything changes

Also, if the provider you are using has been set with Optimize Audio to Yes, set this to No and see if that helps...

I suspect that when calling out, especially if Optimize Audio is set to Yes, the SIP ReInvite is using 5060...hence your issues

Dear emoci,

Thank you for your help.

I've tried ports 80 and 443 and give the same as 2060.

Optimize Audio Path is set to NO
NAT Handling is set to YES.

I get a connection to any provider i have available in my voxalot account, (through my dialing plans), the connection operates flawlessly for 21 seconds and then sudenly disconnects!

I see other users referring to this in previous posts in this same thread.

Any help is apreciated.
Best Regards,
~thanasis.

Ok, how about NAT routing, especially STUN setup:

I am not familiar with the Fritzbox but here is a few things to consider: http://forum.voxalot.com/voxalot-sup...html#post14452

Dear emoci,

Thanks for all the info.

I've purchased a Linksys/Sipura SPA3102 to replace my fritz.box 7140. I have one provider registed with Line1 and Voxalot as my Gateway1. The ports used are 2060. Everything is working as expected, except for the disconnect after 20 seconds, that voxalot connections - suffer.

I've setup the STUN: stun.xten.com, and NAT keep alive on my accounts. Also i have forward ports:
2060 UDP and (RTP) 16384-16482 UDP to my SPA3102 adapter.

I've also followed the aritcle metioned in the previous reply.

Still Voxalot will disconnect after exactly 20 seconds.

Any ideas please?

~thanasis

-Open and forward the following ports to the ATA:

5050-5064 Both TCP/UDP
5000-5005 Both TCP/UDP
16300-16500 Both TCP/UDP
2060 Both TCP/UDP

-Under 'NAT Support Parameters heading' try these:

Handle VIA received: Yes
Handle VIA rport: Yes
Insert VIA received: Yes
Insert VIA rport: Yes
Substitute VIA Addr: No
Send Resp To Src Port: No
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.xten.com
EXT IP: (blank)
EXT RTP Port Min: (blank)
NAT Keep Alive Intvl: 20 (or 30)

or

Handle VIA received: No
Handle VIA rport: No
Insert VIA received: No
Insert VIA rport: No
Substitute VIA Addr: Yes
Send Resp To Src Port: Yes
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.xten.com
EXT IP: (blank)
EXT RTP Port Min: (blank)
NAT Keep Alive Intvl: 20 (or 30)


-Also in your VoXalot acct. try the following combinations:

NAT Symmetric Handling: NO
Optimize Audio: Yes

or

NAT Symmetric Handling: NO
Optimize Audio: NO

or

NAT Symmetric Handling: Yes
Optimize Audio: NO


-Make sure that 'NAT Keep Alive', and 'NAT Mapping Enable' are both set to Yes in the SPA

I have NAT keep alive and NAT mapping on the PAP2 set. When I register to us.voxalot.com:2060, 80, 443 i.e. anything besides 5060 I see that

When I call another voxalot user who is offline (not registered) then the call does not go to his voicemail. I just get no response. I am able to call the user when he is online (registered) and reach his voicemail.

What is the problem here ?

Thank you.

This should not happen according to emoci..... is this problem already resolved ?

This happens within 21 seconds ?

Two questions:

-Does this or does this not happen when you actually use port 5060?
-Does this continue to happen if you call in the format *010xxxxxx rather than just 6 digit dialing?

One thing that maybe occurring, if you are calling another VoXalot user within 5-10 min after their device has turned off, sometimes the servers have not propagated the change yet...so rather than hearing ringing you may receive dead space equivalent in length to the time it would ring... then VoiceMail should pick up (vs. VoiceMail picking up right away when the de-registration is complete)...

I'm also on top of this one, have the same 20 second drop connection, it is NOT the ATA device, but some keep alive function send from Voxalot, that gets lost in the process, also checking (your) IP address. have lots of other accounts that work perfect, only Voxalot, gives this problem.

Wait, did you not solve this by changing out your router?
In which case it was likely your router that was dropping subsequent invites...

Under normal conditions having the following in place should provide for a reliable connection:
-UPnP
-STUN Server
-Correct range of ports open
-NAT Mapping enable and NAT keep alive enabled

Furthermore, most of the users do not have this problem ..in most cases it is examples like your own, where your setup/equipment is interfering with proper functioning, as you found when replacing your router returned things back to working properly

Nonetheless, the reason the forums are here is to run ideas by each other, it is likely that there is another user with similar problems that has a solution, or that in trying all the suggestions here, it will finally fix the incorrect setting causing the issue

Yeah, i hope so, so i don't have to throw my new modem/router into the trash :( never had any problems with any provider or switch service, like i now have with Voxalot, but i don't feel alone any more :)
I did came across a user, who solved the problem, by forwarding 5060, and the range 10000-20000 ports, 5060 is not the problem with me i guess, only to which address should one forward ports ? my "IP address", which can be found tru > run > cmd > ipconfig, ? i forwarded the 10K-20K range also to that address but didn't helped, is there a good guide on port forwarding ?
and DMZ ? to which address should one set this one ? is that safe to do ?
The 21 second connection drop is also the time the betamax provider doesn't "see" that the account owner "shares" it's ip address with other users through a non approved betamax switch service, like MSS.
So naturally, Voxalot registered at MSS has no use, and FUP stays an issue that way, but that way, my connection stays up, but doesn't solve the FUP issue. (sorry for the echo :rolleyes: )examining the sip communication has also no use, since it is a network communication issue..:( that explains also why a softpone doesn't change the situation, only one way audio problems are solved with that, if that's an option... btw. switched off the "N" feature for the wireless part, it only has to with HF signal at 2.4GHz that's 40MHz, instead of the 20MHz in normal "G" mode, so this is not connected to the 21 second connection lost issue.

btw. what we are experiencing, isn't caused by the Premium switch, like suggested elsewhere ?

btw2. what did you mean with my router dropping invites ? i guess my router has nothing to do with the SIP invites, only the the SIP client does, be it a device or softphone i guess,

@OP: after those 21 seconds, is your connection completely lost, or just the audio ?

(MSS=MySipSwitch)

1. When opening ports, they should be forwarded to the internal IP address of your ATA (for the Linksys series I think it is something like 192.168.1.xxx )

2. The fact that switching out the router fixed the problem, likely means that the original router was not forwarding further traffic for the ATA (or the SoftPhone) properly after the first little while (furthermore from the log you provided, Martin concluded that the BYE message was being initiated from your end which is in-line with the above assumption)

In my personal opinion (and I hope others here agree with me) and as I told to you via PM, security-wise I see little to no difference between the WRT54GL vs. the WAG160N....in fact my personal preference is with the WRT54G series just because there is substantial support and firmware from the open source community that provides a rich choice of options beyond what's included with the stock firmware.

Thanks, emoci, i hope also the OP has some usefulness out of this, pointing him also in a good direction.

Using the WRT54GL will mean for me i still have no modem, which is included in the WAG160N, or the old SMC device,
I guess i should use the the WAG160N as modem only, and use DHCP, and the router functions of the WRT54GL.

Oh, I see ADSL2 modem as well...if you are gonna run the WAG160N as a modem, I suggest making sure it is in Bridge Mode, and let your router actually do the login (although this depends on how your system completes the handshake, is it PPPOE, PPPOA or a dedicated line etc.)

Look, frankly I'd suggest using what works for you.... if the SMC you have on right now does not produce any other issues then just leave it in (As they say, Don't Fix what's not broken:) )

If your only worry for the changes is security, a good personal firewall, and a reliable AntiVirus Software should be just fine ....

Yeah i'm starting to believe that too, (the SMC option) it's the easy way out, i also experienced also what i believe, a crash of the WAG160N, it wasn't accessible on it's web server address...(+ one LAN port indicator very fast flashing..)
The modem type is a PPPoA type protocol, the Tomato fw has only PPPoE option.
Well, at least the WAG160N looks pretty, (see avatar)..:(
I guess a Linux router, is the most configurable one, (PC with two ethernet cards) somewhere running in a staircase closet 24h/day :)

I guess Voxalot isn't an option for a lot of people, since a connection through a open wifi network will not work most of the time....

Although there have been some issues for some users, those who have used the nokia tools and have specified the STUN server as a result seem to be doing ok....

See Nokia Configuration Settings - Voxalot FAQ for guidance

The errors i encountered where not NAT related, and lie somewhere in the ADSL layer of the hardware, my ADSL provider could also be the cause, but only because my modem/router is the very new WAG160N, together this results in incompatibilities with the Voxalot service, no other provider or service like MySipSwitch, give me problems like i now have with Voxalot.

Thanks for multiple sip proxy ports.

Can anyone please confirm port 80 and 443? I seem to have trouble with those two and 3060 4060 5060 work fine. Want to make sure if its something from my end. Thanks...

SIP port 5090 for d2.spikko.com
 

Hello,

I am using d2.spikko.com proxy and they require port SIP 5090. I've configured my service provider list with port 5090 and I see status Registered. When I call my number on this I get a regular dialing tone but my telephone (connected to my SPA-3102 does not ring at all and I can not make calls using Spikko provider.

I see SIP ports 80, 443, XX60 are open on voxalot side so I assumed I would not even see Registred status when using 5090.

Does anyone has had experience with Spikko VSP? From some internet threads I found there seems to be some problem with Asterisk servers using port 5090, but then again I do not have enough experience nor information.

BTW, when I configure my soft-phone (SJPhone) using same parameters it does work perfectly.

Any ideas?

Thanks.

I try to use spikko, but I try all combinations, it always gives me this:
spikko, fetching registrations failed: 503 Service Unavailable

so I can`t use it, it is notregistering on twinkle and not on x-lite.
someone can help please?

As far as I can tell Spikko connot be used via SIP at the moment. Having tried myself with VoXalot, MySipSwitch or a SoftPhone...nothing seems to work ...

I realize there are a few articles that claim otherwise...but they are a bit dated...

yes, they write on their site it will work with ATA sip and also on x-lite it is logged in and on sipsorcery too, but I do not know how to make a test call, they made it very complicated to do a call from a regular cellphone, this costs me too much money to try test calls and only the voicemail is answerin,g and then they ask for a number i just do not understand why it is impossible to call direct to the spikkophone number I got from them, even no forum there.

Cannot register my v100 to voxalot was working for years registers when I change to M
 

Cannot register my v100 to voxalot was working for years registers when I change to MNF but no success on the ATA with Voxalot ( is registering on x-lite softphone ? Any problems at voxalot ? any solution

Ta Jak