Request for Make Call Without Register

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What steps will reproduce the problem?
1. Remove Username, Password and Server from Settings -> Siphon 
2. Or VSP server is unreachable
3.

What is the expected output? What do you see instead?
For case 1.) I expect Siphon can start up without VSP credential
information.  What I see is Siphon refused to start up and reported "Siphon
requires a valid SIP account."

For case 2.) I believe Siphon is already able to identify whether it is a
VSP server unreachable case or really invalid username / password being
used. But for the serverr unreachable case Siphon reported: 

 "Invalid value for username or server. To correct this parameter, select
"Settings"..." 

this will confuse the users. I found Siphon is already able to reported
"Service Unavailable" in the upper right side of the Dial Pad. Why still
instruct the user to correct the "Settings"?

What version of the product are you using? On what operating system?
Cydia/APT(A): Siphon (2.2.0-1)

Please provide any additional information below.
The reason I request case 1.) option is due to better support for Direct IP
Dial, I expect Siphon do not need VSP credential when I dail:

 sip://user@1.2.3.4:5061 or sip://user@myATA.dyndns.org:5061

(Remark: I already tested the above sip address, Siphon works for both
address types but I still need to successful registered to a VSP in order
to start up Siphon. As you can see VSP in between is not required in Direct
IP Dial.  Some times it is in the VSP unreachable situation people want to
bypass the VSP and make their Direct IP Dial.

For case 2.) Siphon reported "Invalid value for username or server" upon
start up when VSP is unreachable but Siphon can be started up successful. 

When attempt to dial a number it report the same error message which is not
appropriate for a VSP server unreachable case, 

when attempt to dial a Direct IP Dial sip address from the Contacts Menu,
it simply refused to dial out but this time do not prompt for correct the
value which is good because it will not confuse the user.

Original issue reported on code.google.com by samson...@gmail.com on 27 Feb 2010 at 2:34

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Please ignore case 2.), I found for server unreachable it simply attempt 
"Calling"
without report anything.

Original comment by samson...@gmail.com on 27 Feb 2010 at 2:53

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Original comment by samuelv0...@gmail.com on 27 Feb 2010 at 8:28

• Added labels: Type-Enhancement
• Removed labels: Type-Defect

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I have a good new, just figure out that Siphon can make Direct IP Dial with HTTP
digest authentication enaled in my SPA3102 PSTN GW - although Siphon reported 
in the
upper right above the dial pad that "Not implemented" but it allow me to dail 
out and
the phone no. that I dial passed directly to the PSTN network without making 
any 2nd
stage dialing.

For those who want to try can configure the following:

Settings -> Siphon -> 

Server = myATA.mydomain.org:5061
(include the port or not depends on your implemention)

Password = Yourpassword
Advanced settings -> Auth. Username = Yourusername

the above Yourpassword and Yourusername need to be matched with SPA3102 PSTN 
Line Tab
-> VoIP Users and Passwords (HTTP Authentication) pair.

Original comment by samson...@gmail.com on 1 Mar 2010 at 12:39

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I was wrong, Direct IP dial doesn't work if I test the connection from an 
outside LAN.

Original comment by samson...@gmail.com on 11 Apr 2010 at 4:02

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[deleted comment]

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Connection failure from outside LAN due to NAT problems, can Siphon add better 
NAT
Support to insert and handle VIA received and rport parameters:

1) Handle VIA received : Indicates whether Siphon should react when it 
encounters a
"Via: ...;received=someIPaddr". If set to "yes", then Siphon will use the IP 
address
specified in the "received=" as the public IP address of Siphon.

2) Handle VIA rport : Indicates whether Siphon should react when it encounters a
"Via: ...;rport=someUDPport". If set to "yes", then Siphon will use the UDP port
specified in the "rport=" as the public UDP port of Siphon.

3) Insert VIA received : Indicates whether Siphon should add the SIP header, 
"Via:
...;received=someIPaddr". If set to "yes", then SIP replies from Siphon will 
add a
"Via:" header that reflects the public IP address Siphon saw originating the SIP
request.

4) Insert VIA rport : Indicates whether Siphon should add the SIP header, "Via:
...;rport=someUDPport". If set to "yes", then SIP replies from Siphon will add a
"Via:" header that reflects the public UDP port Siphon saw originating the SIP 
request.

5) Substitute VIA Addr : Indicates whether the public IP address and port 
discovered
by STUN should be used instead of the Siphon current IP address and port (which 
may
be a private address).

6) Send Resp To Src Port : Indicates whether to ignore the "Contact:" header and
instead send the reply directly back to the IP address and port originating the 
SIP
message. 

Original comment by samson...@gmail.com on 4 May 2010 at 4:28