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?
Related
Has anyone been able to configure the Outlook email to pick up email from a Shaw account? (Shaw is an internet provider here in Canada)
I went to the Shaw site and changed the POP3 and SMTP servers to the ones suggested, but I'm still getting a 'not able to download messages' message from Pocket Outlook..
Thanks for your help.
Got it!
OK, I got it to work, so I thought I'd post it here for others, just in case someone else had difficulty.
email address = [email protected]
Incoming Mail Server = pop.shaw.ca (type=POP3 is greyed out)
Username is xxxxx (the first portion of your email address before the @)
Be sure to tick the 'Save Password' box
Outgoing Mail Server = shawmail.cg.shawcable.net
Be sure to untick/clear the 'Outgoing Server requires authentication' box
....then, just set things up as you want them.
Have fun!
drtolson said:
OK, I got it to work, so I thought I'd post it here for others, just in case someone else had difficulty.
email address = [email protected]
Incoming Mail Server = pop.shaw.ca (type=POP3 is greyed out)
Username is xxxxx (the first portion of your email address before the @)
Be sure to tick the 'Save Password' box
Outgoing Mail Server = shawmail.cg.shawcable.net
Be sure to untick/clear the 'Outgoing Server requires authentication' box
....then, just set things up as you want them.
Have fun!
Click to expand...
Click to collapse
Thanks! I don't know why shaw doesn't have this info on their site...Super FAIL by Shaw!
Does anyone know what the IMAP server is? Maybe I will try imap.shaw.ca
Titus_Andronicus said:
Thanks! I don't know why shaw doesn't have this info on their site...Super FAIL by Shaw!
Does anyone know what the IMAP server is? Maybe I will try imap.shaw.ca
Click to expand...
Click to collapse
IMAP servers are all named mail.shawcable.com good luck
I have been struggling with this for almost 3 months now, and never seemed to be able to get it to work properly, sending both on 3G and Wi-fi using the same server information.
Turns out, it was impossible. Shaw just wasn't set-up for it.
However, THEY'VE FIXED IT!
Here is what you need to do.
----------------------------------------------------------
First, you need to goto your shaw webmail and access the new Shaw Webmail BETA they have started.
Login there, and goto the preferences in the top right.
Goto the Mobile Access tab, and enable mobile access (hit enable, type in your password, and select save at the bottom).
*Note, notice the the password requirements under this menu, as although you CAN ENABLE mobile access with a password that does not meet the requirements (perhaps an older one you have not updated, like me) you will NOT be able to login on you phone with it, so you'll have to go change the password in the online help page (which is kind of a hassel, to be honest)
Once it is enabled, these are your server settings.
Type of Account - Pop3
Incoming server: pop.shaw.ca
Username - <your account>@shaw.ca *note, you must add the "@shaw.ca"
Pass - <your password>
Security Type: None
Port: 110
Outgoing Server: mail.shaw.ca
Login is Required.
Username - <your account>@shaw.ca *note, you must add the "@shaw.ca"
Pass - <your pass>
Security Type: TLS
PORT: 587
*the security type and port are not standard... so watch for this.
This will now allow you to both check and send e-mail on both Wifi and 3G.
Just wanted to say thanks for this. Been struggling with this for a few hours. Would never have found the mobile setup preferences in the beta Shaw web mail.
Configuring iPhone4 to use webmail in wifi environment
I recently purchased an iPhone 4 and had it set up for wifi use to access my Shaw Webmail account. Telus is my provider.
My incoming email works fine with the configuration using pop.shaw.ca, my user name, and password.
However, my outgoing has never worked in the Wi-Fi environment. The threads earlier talk about using Shaw Webmail BETA to go to the Mobile Access tab, and enable mobile access.
Shaw has discontinued Shaw Webmail BETA, and have launched Webmail 2.0
so the Webmail site with 2.0 does not have a Mobile Access tab.
Consequently, my outgoing emails under a Wi-Fi environment still won't send.
My outgoing Mail Server was set up by the folks who sold me the iPhone, and it has the following:
SMTP smtp.telus.net
Other servers: shawmail.ok.shawcable.net
ok.shawcable.net
mail.shaw.ca
shaw.cg.shawcable.net
None of the "Other servers" have a username or password inserted. The above other servers were just added a week ago by the retail store, and a message does come up when I'm using wi-fi saying:
"A copy has been placed in your outbox. The recipient "%@" was rejected by the server because the user is unknown."
When i wait awhile and check, the message does send. Anyone know which of the outgoing is the correct one, and is there anything else a person needs to know about Webmail 2.0 to get the settings right??
I just went crazy trying to figure this out for two weeks. Both SHAW and WIND told me it was not possible. I could get the incoming working but not the outgoing.
Here is what worked.
1. SIgn up for shaw beta webmail - type in wmbeta.shaw.ca
2. Under mobile preferences, select ENABLED versus DISABLED.
3. Then the instructions on this PDF work. The forum won't let me paste links, but there is a PDF that comes up from SHAW when you google "shaw email android"
Good luck!
I have the HTC 730 with the Alcaline rom which works like a charm but...
The wifi connection which is configured to connect with the network of our company is a disaster.
We use personal certificates and radius accounting but when I try to connect to the network the device asks me over and over again for my username & domain.
Sometimes I am able to make a connection, but the next day he settings seems to be lost because the problem is exactly the same.
The odd thing is that the username to which the certificate belongs is my window account name (for example ABC1234) en the name which is automaticly filled in the name dialog is my e-mail name whithout the domain suffix.
It has to be my windows account name because thats the only way the connection is created. The only time I am able to create the connection is when my windows name is automaticly filled in and I,am able to push the OK button within a split second.
Is there an application which stores these settings for me? for example the Odyssey client or is it possible to add these settings to the registry or whatever.
I have everything set up correct and Sipdroid works with VOIPBuster and it dials out using my MagicJack trunk, but when I dial, after 30 seconds I hear an error message.
I have all of the information for Username and Password and my proxy is correct. I even changed the proxy to the actual IP address and it still does not work.
Assuming I found my Username & Password and the proxy are there any other tricks I am missing while using SIPDroid?
buddyosher said:
I have everything set up correct and Sipdroid works with VOIPBuster and it dials out using my MagicJack trunk, but when I dial, after 30 seconds I hear an error message.
I have all of the information for Username and Password and my proxy is correct. I even changed the proxy to the actual IP address and it still does not work.
Assuming I found my Username & Password and the proxy are there any other tricks I am missing while using SIPDroid?
Click to expand...
Click to collapse
I read that MJ folks changed to rotating password.. You may have it running until they rotate the password, and by not having the MJ itself, you get logged out. No solution currently, and it was made so to promote magic talk. it sucks because i don't want to make calls from my laptop, and there is no solution for mobile. NetTalk works, but has echo issues, latency is horrible, and it is Wi-Fi only..
So I have a Trixbox server I'm trying to connect to. I've used 3CX, Sipdroid, etc. with no issues but would like to try it native. I add the IP address of the server, my UN and PW and try to make a call. Regardless of whether I'm trying an extension on the box or a regular number, it says "Called from outside...(cuts off)".
Any ideas?
To clarify, I'm referring to the built in SIP client.
same deal with me. i've tried onsip and an internal asterisk server. seem like garbage.
Might be a setting in Asterisk. I ran into this same issue over the holiday weekend and got it fixed with this post:
http://forum.sipsorcery.com/viewtopic.php?f=6&t=3029#p18098
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