Sipbroker doesn't connect to my ATA
 

Hi,

I've been using sipbroker for some time now and I was never able to make a propar PSTN--->Voip call.

Since the last 3 month, I've been using the Quebec City (1-418....) sipbroker gateway to "call" my ATA using my gizmo account. As of today, it always disconect (after 0.1 sec..if you ask me) when I answer my phone on the ATA (or even the gizmo client).

Anyway, I've been trying something new : direct call to my ATA.

I'm now using an ALIAS to point directly to my Ata "voip adress" and the sip broker pstn gateway doesn't seem to work with it.

I've been testing using Gizmocall.com (not beeing logued to my gizmo account) and Calling my own ATA work fine (eg : myUser@myIpAdress) I have 2 way audio.

When I try to call my Sipbroker alias with gizmocall (eg : *12345@sipbroker.com) The path seem to be "ok" but I get 1 way audio...

When I call the Quebec city (LES.net !) pstn gateway, I dial my sipbroker number (eg: *011-12345) and I get a buzy signal... not even a ring on my ATA (and my syslog don't show any activity)

Could someone help me finaly get this to work. I really apreciate the sipbroker service but I never was able to make a call (to my ATA or Gizmo account) that would connect/work right.

Thanks a lot in advance !

anyone could help ?

Sipbroker doesn't work well with Gizmo. Try Point One
 

Sipbroker used to work quite well with Gizmo, but not lately. Not since late last year, at least. Since then, about half my calls through Sipbroker to Gizmo disconnect immediately as you describe, or result in a busy signal. Doesn't seem to matter which Sipbroker PSTN access number I dial into.

It's worse when calling into Gizmo's conference rooms (1-222-xxx-xxxx). In dozens of attempts over the past few months, all have failed in the same way.

Both used to work. I don't know why the service has degraded. I've also reported this on the Gizmo forums, but have received no response there.

Although there aren't as many of them as Sipbroker PSTN gateways, you might try using PointOne PSTN gateways instead. They always work for me.

Same is happening to myself, what's weird is that not all Gizmo # get busy tone from the same gateway. What works is if register with e164.org After you are registered dial the registered # instead to you Gizmo VOIP number. This works and no more busy tone.

I have not had any problems when calling Gizmo numbers through Sipbroker. Just now I made test calls to several 1222xxxxxxx conference numbers. The SIP addresses were formated like this: *7471222xxxxxxx@sipbroker.com