Help with Asterisk and Voicemail Call from MJ Trunk

[DarwinPeru]

Hello: I hope that you can understund my english:
I have my Asterisk Server and I am using many Trunks with MJ Numbers and one Betamax of VoipBuster, that is working perfect, but, when I want call a one telephone number using a MJ Trunk, if the other person hangs Up the call without answer that (or Ignore the call), I hear a Busy sound, and my CLI show that:

-- Called AnyNumber@9730036007
-- SIP/8184336007-0000002a is ringing
-- SIP/8184336007-0000002a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [3056486377@Gerente:22] Hangup("SIP/200-00000029", "") in new stack
== Spawn extension (Gerente, 3056486377, 22) exited non-zero on 'SIP/200-00000029'

But, if when I use my VOIPBUSTER Trunk and the other person ignore the call, that call is sended to the voicemail of the other person, like this, THE NUMBER YOU ARE DIALEGED ########## IS NOT AVAILABLE, PLEASE LEAVE YOUR MESSEGE AFTER THE TONE ..........

and the CLI show that:

-- Called 1AnyNumber@Betamax
-- SIP/Betamax-0000002e is making progress passing it to SIP/200-0000002d
-- SIP/Betamax-0000002e answered SIP/200-0000002d
== Spawn extension (Gerente, 3056486377, 29) exited non-zero on 'SIP/200-0000002d'

it is happening whit any number.

Somebody know why that happens...?

How to I do in my Asterisk Server for call the voicemail of the number called...if the call is ignored?


Thanks

[crackerjack]

Based on the details you are showing, the first call is never connecting to the farside number.

Are you sure you are registering correctly with mj server?


please post details of sip show registry from CLI

[DarwinPeru]

Of course...

it is:

Asterisk*CLI> sip show registry
Host Username Refresh State Reg.Time
proxy1.SanFrancisco.talk4free. Exxxxxxxxxxx0 105 Registered Sun, 07 Feb 2010 17:27:28
proxy1.SanFrancisco.talk4free. Exxxxxxxxxxx0 105 Registered Sun, 07 Feb 2010 17:27:29
proxy1.SanFrancisco.talk4free. Exxxxxxxxxxx0 105 Registered Sun, 07 Feb 2010 17:27:29
proxy1.SanFrancisco.talk4free. Exxxxxxxxxxx0 105 Registered Sun, 07 Feb 2010 17:27:29
proxy1.Miami.talk4free.com:507 Exxxxxxxxxxx0 105 Registered Sun, 07 Feb 2010 17:27:27

[DarwinPeru]

Based on the details you are showing, the first call is never connecting to the farside number

I can make calls normally with my MJ Trunks, my question is about how responds Asterisk when the other person ignore the call when I am using the MJ Trunk and the VOIPBUSTER Trunk....

Thanks