Need some detail logs/diagnostics...
 

I have a problem with Voxalot and a VSP.

I made a call today to someone I was using Netmeeting with.
When I hung up, he couldn't dial out - his line was still tied up. I could dial out though, and confirmed he was showing busy.

Voxalot show my call ending when I put the phone down, and a call duration of 00:25:59.

My VSP show a call duration of 00:38:52 - another 13 mins.

Can we get some logs for the calls so we can check the signals sent from my ATA to Voxalot (which seem ok as it registered call end) and then onto the VSP, which apparently didn't?

My VSP is charging for the call time, so I'm losing out - and would like to find out if this is my ATA needing a kick, or Voxalot, or my VSP....

If the Voxalot call record is reflecting the correct duration then by design a SIP BYE message should be being sent to the upstream VSP.

As you point out, the problem is not with your ATA but rather the Voxalot --> VSP leg.

Is what you describe happening with every call using this particular VSP?

No - but they've answered my query and explained something plausible :

Without talking to the tech team I expect that this is a voxalot/DNS issue -
My reasoning is because I have seen this before where a device was sending
SIP signals to an IP address that it had resolved during a call and not that
which the call had started on. Asterisk suffers from this.

We have multiple servers for redundancy, on multiple IP addresses.
Your device, or Voxalot resolved our address as 1.1.1.1 and places a call -
during the call all is well. However should what ever voxalot is running do
a DNS lookup and resolve 2.2.2.2 for us it 'may' send any SIP messages to
2.2.2.2 which is not where your call is, so your call stays open as it
hasn't been closed.

I'll take this up with the tech team and ask for advice.

For Asterisk the solution is to configure the outbound proxy to be one
specially for this issue as it has only one IP address !
'single-01.proxy.voip.co.uk'

More information is available here:
products:asterisk [wiki.VoIP.co.uk]



So I could change the VSP details in Voxalot, but this removes resilience.

Anything you can do your end? :)

[EDIT]

This has happened again - a 1 min 6 second call stayed open for 4 hours 19 mins and cost me £20 UKP. I've now set my proxy to the one they suggest, and see if this alleviates the problem.

I will discuss with the other techs the possibility of adding an outbound proxy field to the advanced provider details page.

I'm testing with a change to the VSP address, using the single host, rather than the name they normally use (which obviously points to multiple machines) - so FAR it's behaving.

Does SIP not respond with an actual machine (host) name? i.e. if I connect to 'main.vsp.com' and they pass the call to 'box1.vsp.com', can you detect this and 'fix' the call to it?

This is still happening, but interestingly only (appears to) effect 08** freephone numbers.

Voxalot logs show the call ending, so my ATA is signalling Voxalot. I've set the proxy (as recommended) on the Voxalot end, so:

a) Either Voxalot isn't sending a SIP BYE to my VSP
b) my VSP isn't getting a SIP BYE
c) something else...

Clues?

They've run a SIP trace and there is no sign of a SIP BYE.

Can I send someone the logs via PM? I don't want to broadcast all my details all over the forum :)

*bump* *bump*

BUMP.

There is a problem here, as this only happens when I route the call via Voxalot. ATA to VSP direct (via a dialplan) doesnt do it...