[asterisk-users] SIP Registry Problems

[Brent Vrieze]

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.