Sipdroid+PBXes+Computer running SipToSis and Skype = Awesome

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Jun 7, 2008
14
8
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://xdaforums.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:yourusername-200@pbxes.com:5060 (example: *,sip:McAwesome-200@pbxes.com: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:skypests@127.0.0.1:5070
#from_url="skypests" <sip:5611111111@127.0.0.1:5070>
#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:McAwesome@127.0.0.1:5060>
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.
 

dawiebe

Member
Nov 5, 2009
12
0
Thanks leetlikeawping for this how-to.

Where did the name "spiersad" come from? is it your skype username?
 

agentkalaw

Senior Member
Jul 22, 2009
128
11
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?
 

carlostmc

New member
Oct 17, 2009
4
0
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!!)
 
J

jeremybara

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

Dstevens0074

New member
Sep 7, 2010
1
0
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 dstevens004@gmail.com to advise.

Thank you very much.
 

atoschev

Member
Oct 28, 2009
5
0
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:skype@127.0.0.1:5060
from_url="skype" <sip:McAwesome-222@pbxes.org: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:200@pbxes.com:5060) for routing incomung skype calls.

That's all. It really works.
 

bsaurette

New member
Nov 25, 2010
1
0
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:skype@127.0.0.1:5060

from_url="skype" <sip:McAwesome-222@pbxes.org: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 bsaurette@gmail.com

Thanks a ton
 

DeKrasik

New member
Oct 28, 2010
2
0
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!
 

DeKrasik

New member
Oct 28, 2010
2
0
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!
 
Last edited:

andTab

Senior Member
Nov 20, 2011
691
90
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?