asterisk users - Dec 2008 - SIP Registry Problems

Having big problems and for months.  Our service provider (via:talk) 
says they are Asterisk friendly but they are not.  Here are the 
specifics (please read the bottom of the msg too)

System:  Dell SM Business server  2GB RAM, Core II Processor  (should be 
plenty)
OS:  open SUSE 11
Asterisk Version: 1.4.2
Asterisk GUI Version: 2.0

The system was completely set up using the Asterisk GUI with a couple 
tweaks in users.conf that via:talk wants.

Here is what happens:
1.  Asterisk verifies connection to the server and we get this.  (CLI 
output)
    -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
mapped to host galvatron.vtnoc.net, port 5060
    -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
mapped to host optimusprime.vtnoc.net, port 5060
    -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
mapped to host megatron.vtnoc.net, port 5060
    It jumps around from server to server all the time.

2.  With all the server jumping sometimes incoming calls get re-routed 
by via:talk to the bosses cell phone, the fail safe dump off number.  
Seconds after calling and getting re-routed to the boss I call and it 
goes through.

3.  We cannot recieve DTMF from via;talk, have tried auto, rfc2833, and 
inband without success with any of them, and yes we had via:talk change 
their end too.

Here is the users.conf entry or the connection to via:talk.

[trunk_1]
context = DID_trunk_1
host = galvatron.vtnoc.net
username = <phone number>
secret = blablabla
trunkname = via:talk ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
fromuser = <phone number>
authuser = <phone number>
insecure = port,invite
dtmf = inband
dtmfmode = inband
relaxdtmf = yes
;rfc2833compensate = yes
port = 5060
canreinvite = no
disallow = all
allow = ulaw,gsm

I did set up a very basic Asterisk box yesterday that put all the 
conection settings in sip.conf and I even renamed users.conf so it could 
not load.  I then put in about a 10 line hand coded dial plan in 
extensions.conf and got the same results

Of course via:talk is of no help as they only officaly support the 
Linksys PAP2 they sent us with our account.

My solution is to move away form via:talk and leave the problem behind.  
I then figured smeer nasty things on the internet about them but I'm too 
late, many other people already have.  :)  The problem with moving is we 
paid for a years service and that is up in April and the boss is cheap, 
cheap, cheap.

Please help my connection woes and thanks in advance.

Brent Vrieze schrieb:
> Here is what happens:
> 1.  Asterisk verifies connection to the server and we get this.  (CLI 
> output)
>     -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
> mapped to host galvatron.vtnoc.net, port 5060
>     -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
> mapped to host optimusprime.vtnoc.net, port 5060
>     -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' 
> mapped to host megatron.vtnoc.net, port 5060
>     It jumps around from server to server all the time.
hello,

you should add the second and third server to your user.conf as a friend
too, they just use something like a load balancer but you only accept
calls from one of their 3 servers.

so i think what happens is that if a call comes from one of the server
you didnt authorize it just get an error like "authorization required"
and then fallback to your boss cell number.

maybe you could trace this with sip debug.

best regards.

steve