Database Develop

Public IP Address??

I am trying to run one of the many FTP servers available for Windows Mobile (vxFTPSrv or ShareIT FTP) to keep some files in sync but I can't seem to figure out what public IP address my phone has. vxFTPSrv says it is listening to a non-routable 10.**** while whatismy ip says it starts with a 200.**** while DynDNS for Windows Mobile says it is 32.****. Nevertheless, none of these work nor can I get these programs to listen to the ports. Any ideas? Does the Tilt even get a public IP address from AT&T? Thanks.

Try this
I'm no pro, but had to tell the Physical address to the the tech guy at my university so he could enable my phone o use the wireless network...
... anyways, download and install a Registry Editor (I use Task Manager v2.8) you should be able to access an Ipconfig tab, where all the information is available.
I don't know if I'm in the right track.... Hope this helps? =)
jim

your pda have an ip address
using wifi router the uplink sees the routers ip
using an isp the internet sees the isp's assigned ip
http://www.ip-adress.com/
http://en.wikipedia.org/wiki/IP_address

Here is a kewl utility that is FREE.
http://www.cam.com/vxutil_pers.html
Here is the stuff it does.
DNS Audit
DNS Lookup
Finger
Get HTML
Info
IP Subnet Calculator
Password Generator
Ping
Ping Sweep
Port Scanner
Quote
Time Service
Trace Route
Wake On LAN
Whois
Another good one I use is Iper Suite.
http://tonaya.com/products/iper/index.php
For casual use the first one is probably satisfactory.
IPer is worth buying for the increased functionality and has a TFTP client.
HTH

TSoma said:
I am trying to run one of the many FTP servers available for Windows Mobile (vxFTPSrv or ShareIT FTP) to keep some files in sync but I can't seem to figure out what public IP address my phone has. vxFTPSrv says it is listening to a non-routable 10.**** while whatismy ip says it starts with a 200.**** while DynDNS for Windows Mobile says it is 32.****. Nevertheless, none of these work nor can I get these programs to listen to the ports. Any ideas? Does the Tilt even get a public IP address from AT&T? Thanks.
Click to expand...
Click to collapse
200.* would be a public IP. As would 32.*
Turn off your proxy setting in the phone and do the www.whatismyip.com thing. That will get you a more meaningfull result than anything else.
When I'm on GPRS/Edge (AT&T proxy settings in force) network I get a 66.102.186.15 IP address as reported by whatismyip.com. That resolves to alpmagr1fe06-dmz.mycingular.net. Which should be a att proxy server even tho it implies by its name its on a DMZ.
When I turn off the proxy for GPRS/Edge I get 166.195.188.15 according to whatismyip.com. That IP address will not respond to ICMP commands. So I assume it is firewalled. So it seems to me that yes you CAN and DO get a public IP address, its just that address is heavily firewalled.
You've piqued my interest, tho I cannot do anymore testing at this moment.

Changing Messaging ports

Has anyone found a way to configure messaging to use different SMTP ports than the default TCP/25? I've tried searching the registry, to no apparent luck (although I may have missed something).
We run our SSL IMAP on TCP/465, and with the "enable ssl for outgoing" it still tries to connect to 25.
This is on a hideously hacked ROM based on Alex.v5.ATT.161.UltraClean.ROM, FWIW.

How are you currently trying to assign ports? I don't remember if I've tried this on a PPC it like the following:
imap.servername.net:465

PsyOpWarlord said:
How are you currently trying to assign ports? I don't remember if I've tried this on a PPC it like the following:
imap.servername.net:465
Click to expand...
Click to collapse
Yep, tried that. Blows it off completely. I even tried using IPort rather than DNS Nameort.
For the life of me, it seems that SMTP is harcoded at 25 regardless of the SSL flag.

Unfortunately I'm not sure. I don't have an IMAP server to test on right now.

Email connection problem

Hi
I've been trying to setup an IMAP account on the built in email. But can't seem to connect to it, it says:
"Cannot connect to the incoming mail server. Be certain that your login information is correct, then try again to send and receive email"
But I'm definitely sure all email account info is correct.
Wierd thing is, when I go back to the email Accounts setup, I notice the password there rather than ******* 7 characters long password it is like ********************** 20 characters long. I've tried changing back to the real password, but still don't work and if I check the Accounts setup again, the password, again shows really long like ***********************
I've tried accessing my IMAP email account through the internet on the phone via Opera, and that's working OK.
Any ideas?

Mmmm...
Can't really help you with your settings, but I can answer one question for you. The masked password is always a long string of ****'s. This is so no one can guess the actual number of characters in your password.
I've never set up an IMAP account, as I use POP3. I had an issue years ago when I was first trying to figure out how to set email up on these things, but I'd just got the incoming pop address wrong. I went on the internet and Googled the proper settings. Sorry I can't be of any more help

During the email set up there is a link for Advanced Server Settings. There there are settings for Require SSL for Incoming/Outgoing email. You might try toggling those. It depends on the IMAP server setup. Which IMAP server are you trying to connect to? I run my own IMAP server. The difference between IMAP and IMAPS (i.e. with SSL) is that you connect on a different port. If your IMAP server only accepts IMAPS then it's listening on one port but you're asking on another.

I've noticed that the email doesn't work when using activesync for network, but it's fine if using WiFi for example.
Actually, the problem is probably Activesync itself, as I have network problems with it... Activesync seems unable to remember proxy settings, its like it works after the initial setup , for a few minutes, then any proxies that were setup there are gone a few minutes later, just dissappears and consequently the network won't work anymore until I manually set it up again

Sipdroid+PBXes+Computer running SipToSis and Skype = Awesome

First post ever! I have been a leecher for a good year now and have marveled at all the incredible stuff that you people post in these forums. I hope that this will help some of you out, if at least a few. First of all I would like to thank gurnted, if it wasn't for him I wouldn't have spent a whole day researching how to make this work. If you haven't read his post yet I would highly recommend it. http://forum.xda-developers.com/showthread.php?t=548405
Anyways on to the meat and potatoes of the post. This is a guide to setup incoming and outgoing skype calls via your wifi or 3g networks.
Things you will need:
-Skype account with latest client. (and preferably some kind of subscription)
-PBXes account.
-Computer running SipToSis software http://www.brothersoft.com/siptosis-295109.html and skype client. (need to be running pretty much all the time, or atleast whenever you want to make or recieve calls)
-Sipdroid app.
First thing is first. Create an account at PBXes.com https://www1.pbxes.com/index.php Log into the account and go to extensions. Click SIP then under extension number type 200 then click submit. Next go to Ring groups. Type 00 next to extension list, 60 next to ring time, and check next to hangup then click submit. Next click add ringroup. This time type 200 next to extension, 60 again at ring time and check next to hangup again. Submit that and then click on trunks. Click add sip trunk. Next to sip server type the wan ip address of your router or whatever ip address your ISP gave you( personally I use a DynDns service http://www.dyndns.com/ that is updated via my DD-WRT router. You can use this sight to find your IP http://ping.eu/). You can if you like put a username and password here but I havent figured out how make the SipToSis script ask for my username and password yet. Anyways, give this trunk a name before you go(can be anything) then click submit.
Ok breather for one sec. personal note learn to use paragraphs.
Ok check next to ring group and select 1 for both regular hours and after hours. Next put an asterisk next to regular hours and days at every line(not sure if this is necessary). Click submit and then click add-incoming route. This time next to trunk type your username that you used to login and the -200 (example is if your username was McAwesome then type McAwesome-200). Choose ring group again for both regular hours and after hours but this time choose 2 in the pulldown. Once again put an asterisk next to all the regular hours and days. Submit. Next click outbound routing. Route name: put whatever you like. At the pulldown next to trunk sequence choose the trunk you created and then click submit. That does it for the PBXes account, by the way if you see the red bar across the top that says submit changes then go ahead and click that bad boy away.
Download the skype client and install on your computer. Next download the SipToSis software and unzip it to a folder in your favorite directory. Now go to that folder and edit SkypeToSipAuth.props (personally I use notepad++ to do all my editing http://notepad-plus.sourceforge.net/uk/site.htm). At the very bottom edit the line to look like this *,sip:[email protected]:5060 (example: *,sip:[email protected]:5060). Save and close that. Next edit SipToSkypeAuth.props and change the bottom line to look like this *,*,*,calleeid then save it and close. Alright now open siptosis.cfg for edit.
This is the tricky part. Edit these lines
#Sample AUTO config with NO registration
# username and password not important in this mode
# Set to available port to transport SIP messages on siptosis computer
host_port=5070
username=skypests
passwd=unimportantpassword
do_register=no
# --- end of NO registration example ---
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
#host_port=5070
#contact_url=sip:[email protected]:5070
#from_url="skypests" <sip:[email protected]:5070>
#username=skypests
#passwd=unimportantpassword
#realm=127.0.0.1
# --- end of NO registration example ---
To this. Make sure to notice the usename and the port 5060 (I will use McAwesome as an example again.) Also you can put the username and password that you made when you created your trunk but I havent been able to get it to actually ask for the password yet.
#Sample AUTO config with NO registration
# username and password not important in this mode
# Set to available port to transport SIP messages on siptosis computer
#host_port=5070
#username=skypests
#passwd=unimportantpassword
#do_register=no
# --- end of NO registration example ---
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
host_port=5060
contact_url=sip:127.0.0.1:5060
from_url="spiersad" <sip:[email protected]:5060>
username=McAwesome
passwd=yourpassword
realm=127.0.0.1
# --- end of NO registration example ---
Now save and close. Make sure skype is running and execute SipToSis_win.bat. As long as everything went well you will be looking at a cmd window with a bunch of information about skype on it and Ips and all sorts of stuff. Check skype and accept any plugins or whatnot it trys to run.
Now setup your sipdroid app and you will be all set. click the sipdroid app and click menu then go to settings. Click Sip account settings. Put username as your username-200 (once again, example: McAwesome-200). Password is your login password for PBXes account, Server is PBXes.com, Port is 5060, and Protocol is UDP. Go back then go to call options. Check Wlan and 3G if you use it. I set my preferred call type to Sipdroid when available but once again that is your choice. Thats pretty much it for the Sipdroid app.
And thats pretty much it. you can make calls out of sipdroid to your skype client and then worldwide (if you have the supscription). Also any call skype recieves you will recieve.
Couple of last things. you might have to run the SiptoSis script once before you actually start editing it. Also make sure that your windows firewall and router isn't blocking port 5060. I also had a problem where i had to turn off the sip algorithm in the router to get this to work over my wlan.
Ok thanks for reading and hit me up if you have any problems. I will try and get back to you as soon as I can but its kinda rough when you are deployed over seas.

You should get your siptosis program from the auther's site http://www.mhspot.com/sts or from cnet: http://download.cnet.com/SipToSis/3000-2349_4-10969407.html
Brothersoft is running a scam claiming it costs $2.50 - it is not true. You can get it for free from the locations I mentioned. You also run the risk of getting a virus or spyware by not getting software from trusted locations.

Thanks leetlikeawping for this how-to.
Where did the name "spiersad" come from? is it your skype username?

Sounds good!
How is the call quality? I have sipdroid running on gizmo 5. and it works quite well but i can't increase the volume because of echoing so it can get annoying when talking. so is this a sipdroid bug or is it better with pbx since it's designed for it?

I can't seem to get it to work, i find the instructions kind of complicated, do you think you can go over the process again, (some pictures would be great!!)

Thanks so much for the informative post! I was able to get this up and running successfully

I could use some help...
I used you instructions for setting up sipdroid,with sip2sip, on skype. I think I may have done something wrong though.
When i try to make a call from my sipdroid I always get your call can not be completed as dialed no matter if if it is a local or international number.
I was a bit confused about the steps after you had us take a breather as well. I was not sure were to be putting in that information.
So as it stands I have my pbexes set up with:
One extension,
2 ring groups,
1 trunk
2 inbound routing, and
one outbound.
I have version fios and got my wan ip set in the files you had us edit.
I m not sure if I was supposed to change the username and password info under the skype info to mine though.
Feel free to send me an email at [email protected] to advise.
Thank you very much.

one more config issue
Very helpful post!
I've found another way to make same things with new sub pbxes functionality introduced recently.
First of all, you don't needed to create trunc (and collect your external ip) any more. Instead, simply add sub pbxes (e.g. 222) with name/password and choose this name (McAwesome-222) in outbound routing as trunc name in your trunc sequence field.
Change your siptosis settings like this:
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
host_port=5060
contact_url=sip:[email protected]:5060
from_url="skype" <sip:[email protected]:5060>
username=McAwesome #sub_pbxes name
passwd=yourpassword #sub_pbxes password if any
realm=pbxes.org
expires=3600
minregrenewtime=120
regfailretrytime=15
do_register=yes
# --- end of NO registration example ---
Don't forget to modify SkypeToSipAuth.props (in your example: *,sip:[email protected]:5060) for routing incomung skype calls.
That's all. It really works.

Hope someone is still on this thread and able to help.
When I attempt to use the sub PBX method using the following .cfg excerpt
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
host_port=5060
contact_url=sip:[email protected]:5060
from_url="skype" <sip:[email protected]:5060>
username=***** #sub_pbxes name
passwd=***** #sub_pbxes password if any
realm=pbxes.org
expires=3600
minregrenewtime=120
regfailretrytime=15
do_register=yes
# --- end of NO registration example ----
I am unable to register on siptosis when I run it due to time-out. Sipdroid has no problem registering.
When I use the trunck and ip address method I can register both my phone and siptosis but when I make a call I get a message saying your call cannot be completed as dialed.
Also for the auto config, is it supposed to be 5060? (I would guess it doesn't matter as its not going to connect anyway)
any suggestions, I'm reachable at [email protected]
Thanks a ton

Thanks for the tutorial. Works like a charm! Was a bit frustrating at first but not it works. Can't believe this is possible. Free calls over any wifi! I'm using my brother's skype account to which he has a subscription. I can click any contact in my phone and it routes it through sipdroid instantly. Beautiful!

I can't seem to get incoming calls. How exactly do I go about setting that up? I have an extension with 200 set up and an Incoming Route set up. Followed the instructions but to no avail. Thanks for any help!

Sorry for reviving such an old thread but this seems to be one of the most knowledgeable resources about what I am trying to achieve:
1. I have a Google Voice account and a free DID from callcentric.com
2. The GV account is set up with the free DID, so calls to my GV forward to the DID and it is possible to trigger a call from the website (callback to the DID and connecting the outgoing call once I pick up).
3. I am using a real SIP phone as my device at home (Linksys SPA942) with the free DID.
4. I also have a Skype plan with a dial in number that would allow me to make free outgoing calls if I could get the SIP phone to trigger a connection to that dial-in number.
Is there a way to trigger an automatic callback from the SIP phone (either to my fixed Skype dial-in, or dynamically connecting via GV), or have my SIP phone connect to Skype for outgoing calls?

[Q] Native SIP calling with Ekiga.net?

Has anyone gotten the native SIP client to work with an ekiga.net account?
I added my account in Call Settigs -> Internet Calling -> Accounts, but when I call the echo test I get nothing.
CSipSimple and sipdroid both work, but I was hoping to use the native stuff...
I'm running stock 2.3.4

I'm also having problems with this. I'm on 2.3.3. When I click the "Receive incoming calls" box, the status of my Ekiga account changes to: Account registration failed: (Not Acceptable(606)); will try later. It tries later, but never gets anywhere.

Same problem here.

I wonder if its because it doesn't support STUN. There's nowhere in the account settings to enter the STUN server.
When I call the echo test number it looks like the call connects up fine (doesn't timeout) but I don't get any audio. Looking through the logs, it also looks like it connects ok.

Seems there is something STUN related not being there in the native SIP client on Android, even in Nougat (Android 7.1.1)...
I got this info from https://code.google.com/p/android/issues/detail?id=15685 Which states that STUN could not be needed IF EKIGA changed some settings on the server side.
I quote:
"#13 jens.mar.. @googlemail.com
Normally, if the voip device uses the "rport" parameter in the sip VIA header, the receiving SIP server sends back the data to IP address and port where the request came from. This way the communication is the same as with an http server. Then you don't need a STUN server.
But if the SIP server of the provider takes the internal IP address from the SIP request, the call initiation will fail. Then you need a STUN server."
So, it seems this could be solved (also), on the EKIGA side, and this may be the way to go as the native Android SIP client works in many other SIP services, except, sad to say, in EKIGA.net