Problem calling another Voxalot

[hddlab]

Hello
I have a Voxalot account and I have a fried that have another, we are having problems when I call him, sometimes the call shutdown with 30 seconds, another time does not conect, others conect but I can hear hin and he ca not hear me.
Some times when I use *010 and I call him, the call goes direct to voicemail.

When we use Gizmo and we call eachothers him using Gizmo and I using Voxalot we do not have this porblems also when we change- he with Voxalot an I using Gizmo same the connection is working.

We already did a lot of tests, and using all the Codecs, and we are getting same result all the time.

Someone here is facing something similar?
Someone here has any ideia of what can be?

Thank you very much for your attention and time

Jose Pinto

[doctorh]

Originally Posted by hddlab

Hello
I have a Voxalot account and I have a fried that have another, we are having problems when I call him, sometimes the call shutdown with 30 seconds, another time does not conect, others conect but I can hear hin and he ca not hear me.
Some times when I use *010 and I call him, the call goes direct to voicemail.

When we use Gizmo and we call eachothers him using Gizmo and I using Voxalot we do not have this porblems also when we change- he with Voxalot an I using Gizmo same the connection is working.

We already did a lot of tests, and using all the Codecs, and we are getting same result all the time.

Someone here is facing something similar?
Someone here has any ideia of what can be?

Thank you very much for your attention and time

Jose Pinto

Yes. This is the same problem reported in other posts, about a service going bad in one of the US proxies. Hopefully can be fixed soon, it started last wednesday...

[martin]

Originally Posted by doctorh

Yes. This is the same problem reported in other posts, about a service going bad in one of the US proxies. Hopefully can be fixed soon, it started last wednesday...

Please see http://forum.voxalot.com/voxalot-sup...html#post24577

[hddlab]

Hi all,
First of all, thank you very much for answers.
The hardware that I´m using: ATA 486 Grandstream and also a softphone Eyebeam, both same result.
Sometimes you can talk others you can´t. After you call another user you also can do one thing change the provider to another that can do peering link Gizmo and you just call the other Voxalot use and works.

I already did many tests, I call the other user direct, I call using the sip code *010, I call using ulaw codec only and I did with all codecs.
And I get same result in my tests.

Also I read here that people that pay has one server and people that does not pay has another, this is something that can be the problem?

Ok, I just would like to say here that this is a problem that the guys here will fix, and I will start to pay for the service anyway. I think that Sipbroker and Voxalot are one of the best things that happens in voice over IP since the first beguin.

Thank you all for your attention and time and alos for allways give us your help.

Jose Pinto
Brazil

[scaev]

Hello, it seems I have the problems mentioned here, too.
I have two Voxalot numbers,
840146 : ATA Planet VIP-156 -> DI-824VUP (full cone NAT) -> public IP address
471473 : SJPhone 1.65 -> DI-824VUP (full cone NAT) -> public IP address
When I call from 840146 through any VSP, voice goes in both directions and everything is OK.
But when I call voxalot-to-voxalot, that is
from 840146 to 471473 or
from 471473 to 840146,
I have either one-way audio or no audio at all. Signalling (ringing, hanging up etc) is OK.
For both numbers,
the echo test (*600) works fine,
STUN is activated in clients,
"Symmetric NAT handling" option is set to "Yes" (if set to "No", nothing is working).
It does not matter if I use eu.voxalot.com or us.voxalot.com.
May I hope that this problem be solved and how?..

[boatman]

Originally Posted by scaev

..."Symmetric NAT handling" option is set to "Yes" (if set to "No", nothing is working).

As far as I know, Voxalot's optional "Symmetric NAT handling" is a feature that can be activated only when there is a fault that causes a non-routable LAN IP address to be used as the Contact Address in a SIP SDP packet. This can happen when the ATA is behind a symmetric NAT router, the ATA is mis-configured, or a STUN server is not working, etc. The Contact Address is supposed to be your public IP address.

Strangely, your Voxalot to Voxalot calls are being routed through SipBroker. This is not normal. Below, part of a SIP packet captured from your incoming test call shows the servers which handled the SIP packet. Note that sipbroker (64.34.162.221) is the second hop for the call.

Record-Route: <sip:64.34.173.199;lr=on;ftag=as0461a5eb>
Record-Route: <sip:72.51.47.59;lr=on;ftag=as0461a5eb>
Record-Route: <sip:64.34.162.221;lr=on;ftag=as0461a5eb>
Via: SIP/2.0/UDP 64.34.173.199;branch=z9hG4bK2b69.a30ed691.0
Via: SIP/2.0/UDP 72.51.47.59;branch=z9hG4bK2b69.6d3f4d22.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bK2b69.eeddfa11.0
Via: SIP/2.0/UDP 64.34.173.199:5061;branch=z9hG4bK5197bd11;rport=50 61
From: "840146" <sip:840146@64.34.173.199:5061>;tag=as0461a5eb
To: <sip:*010608019@sipbroker.com>
Contact: <sip:840146@64.34.173.199:5061>
Call-ID: 6be93ba43f255ff234fdf3977a9e2c54@64.34.173.199
CSeq: 102 INVITE
User-Agent: voxaLot
Max-Forwards: 67

[boatman]

Originally Posted by hddlab

Someone here has any idea of what can be?

What kind of SIP hardware or software are you using? If you send me your Voxalot number I can call you and check your SIP SDP packet for clues to the problem. The call only takes a few seconds, just answer then immediately hang up.