hudamalmsteen / csipsimple_test

Automatically exported from code.google.com/p/csipsimple
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Can someone post a quickstart guide or explain the different settings boxes for advanced account setup? #162

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
What steps will reproduce the problem?
1.  I am trying to get this set up with my pbxes account
2.  Can someone here at least give explanations/definitions for the different 
input boxes under the account settings? 
3.  I am a newbie to SIP/voip systems and these input boxers vary from program 
to program.  
4.  I use pbxes because I wanted an incoming local SIP number...

What is the expected output? What do you see instead?

NA

What version of the product are you using? On what operating system?

NA

Please provide any additional information below.

I am not looking for help as such, just definitions or better explanations of 
the input boxes for advanced account setup...

Original issue reported on code.google.com by jerald...@gmail.com on 24 Aug 2010 at 11:16

GoogleCodeExporter commented 9 years ago
If I manage to figure it out on my own, I'll post it myself :)

Original comment by jerald...@gmail.com on 24 Aug 2010 at 11:17

GoogleCodeExporter commented 9 years ago
I wrote a HOWTO at my blog, you can refer to it 
http://samiux.blogspot.com/2010/08/howto-voice-over-ip-voip-on-android_23.html

Original comment by runner...@gmail.com on 24 Aug 2010 at 11:52

GoogleCodeExporter commented 9 years ago
I got this from http://samiux.blogspot.com/

This is for basic account on pbxes.org, but isn't working for me :(

(A) Account
"Add account" -- "Basic" 

"Account name" - any name you like, e.g. pbxes.org
"User" - username-<extension>, e.g. android-100
"Server" - pbxes.org
"Password" - password 

(B) Settings
(a) Easy Configuration
Nothing to set now.

(b) Network
Check all items.

(c) Media
Check "Echo cancellation"
Check "Voice audio detection"

(d) User interface
Check "Dialer integration"
Check "Text dialer"
Check "Integrate with Music application"
Check "Keep awake while on call"
Check "Use partial wake lock"

Original comment by jerald...@gmail.com on 24 Aug 2010 at 11:59

GoogleCodeExporter commented 9 years ago
PBXES is correctly set up, I have been making calls from it with another app...

Settings are as follows:

Account Name
me-100

User
me-100

Server
pbxes.org

Password
mypassword

this is my current config, with different account name for my privacy ;)

Original comment by jerald...@gmail.com on 25 Aug 2010 at 12:03

GoogleCodeExporter commented 9 years ago
You just dial out in the following format.  Beware that "me" is the account 
name (i.e. login name) in PBXes.org and "100" is the extension.

me-100@pbxes.org

or

me@pbxes.org

Original comment by runner...@gmail.com on 25 Aug 2010 at 3:01

GoogleCodeExporter commented 9 years ago
I'm not sure for me username-<extension> as username. I forgot how pbxes.org 
manage that.

Just an important point, I don't know how your account is configured, but 
pbxes.org is in most case not needed if you have another SIP provider. Sipdroid 
is linked to the company that provide pbxes.org and so all their tutorials link 
pbxes.org, but SIP is not available only with pbxes.org and in most case they 
explain how to use pbxes.org as a SIP proxy and make trunk with another SIP 
account.
But CSipSimple can works directly with the other SIP account.

However, I should probably add a pbxes.org wizard with labels that match the 
pbxes.org interface. Maybe it can helps.

@jeralbsib : is pbxes.org your only sip provider or did you use it with a trunk 
added? If in your first post, by pbxes you mean your own pbx or another 
provider, let me know the configuration you use on any sip softphone (or 
android app) and I'll try to convert it into how to fill either basic or expert 
account according to the complexity of the configuration you need.

@runnersame : your tutorial is really nice, but there is a little confusion I 
think about what is an Asterisk server : sip2sip.info sip server is also an 
Asterisk server. 
Asterisk is just a sip server, as Openser, or others (it's like if you compare 
apache & lighttpd etc). The difference can be the features provided by your 
provider, but not the fact it is an Asterisk or not. Some sip server can be 
media gateways, sip proxy, provide stun...

As you tested and reported to me, sip2sip account can be directly configured. 
We have to find out why in certain condition it automatically hangup but I'll 
send you a mail today if I can get some time to.

Btw, there is a big lack of documentation for now, but as the project is still 
in *alpha* interface will change a lot ! So I fear that docs you'll write right 
now will be outdated for the beta. However, if you are interested in writing 
piece of docs, I can open you the right on the wiki of this website. 

Original comment by r3gis...@gmail.com on 25 Aug 2010 at 12:28

GoogleCodeExporter commented 9 years ago
I understand that this project is in alpha, but it is already one of the better 
SIP apps available with the setup and user interface. 

My PBXES.org account is set up with an outgoing trunk with 12voip as the 
outgoing provider and a DID incoming number with another provider for my local 
phone number with mydivert.com.  I did this because 12voip doesn't provide 
incoming numbers where I am currently, and pbxes was recommended by sipdroid.  
I have it working with pbxes as follows:

@Pbxes.org server:

Setup:

Welcome
People can reach your PBX by calling me@pbxes.org
____________________________________________________________

Extensions:

SIP Extension: 100

Delete Extension 100

Edit Extension

Display Name:   me-100

Webcall

URL:    http://pbxes.org/  
Text:   
Image:  
Latitude:   
Longitude:  

Device Options

username     me-100
password    PassW0R|)
language    English
dtmfmode    Auto
audio bypass    No
dial            SIP/jeraldsib-100

Options

Outbound CID:   

Call Forwarding

All Calls:  
If No Answer:    after 20 seconds
If Unavailable: 
If Busy:    
Call Forking:   
Call Waiting:   

Voicemail & Directory:         
_________________________________________________

Ring Group:

Add Ring Group

group number:   1
ring strategy:  ringall
extension list: 

CID name prefix:    
ring time (max 60 sec): 

Webcall

URL:    http://pbxes.org/  
Text:   
Image:  

Destination if no answer: 

 Extension: me-100 <100>
 SIP URI:   
 Hangup
__________________________________________________

I have 2 trunks, set up in this order:

Edit SIP Trunk

Delete Trunk 12voip

In use by 1 route

General Settings

Trunk Name: 12voip
language:   English
dtmfmode:   Auto
audio bypass:   No

Account

username:   me
password:   PassW0r|>
SIP server: sip.12voip.com
register:   no (just outbound calls

Options

Outbound Caller ID: 31123456789
Maximum channels:   
Maximum outbound channels:  

Dial Rules

Dial Rules: 
Outbound Dial Prefix:   

________________________________________________________________

Edit SIP Trunk

Delete Trunk sip.mydivert.com

In use by 1 route

General Settings

Trunk Name: sip.mydivert.com
language:   English
dtmfmode:   Auto
audio bypass:   No

Account

username:   001101234
password:   abcdef
SIP server: sip.mydivert.com
register:   Yes (inbound and outbound calls)

Options

Outbound Caller ID: 31123456789
Maximum channels:   
Maximum outbound channels:  

Dial Rules

Dial Rules: 
Outbound Dial Prefix:   
____________________________________________________________________________

Route: /

Delete Route /

Edit Incoming Route

Trunk:  
Caller ID Number:   

Set Destination

Regular Hours   
 Extension: me-100 <100>
 SIP URI:   
 Hangup
Special Services
 Callthru PIN:  
After Hours
 Extension: me-100 <100>
 SIP URI:   
 Hangup
Regular Hours:  
Days:   
Regular Hours:  
Days:   
Regular Hours:  
Days:   
 No override (obey the above settings)  (here i have the radio button selected)
 Force regular hours
 Force after hours

Options

CID name prefix:    
Privacy Manager:    No

__________________________________________________________________

Outbound routing:

Edit Route

Delete Route local

Edit Route

Route Name:  local   
Trunk Sequence:
0             SIP/12voip

Set Destination

 Valid for all numbers    (radio button here is selected)

 Numbers starting by: 
     Separate prefix    

 Custom Dial Patterns:  

Options

Route Password: 
Extension:  
____________________________________________________________________

Edit Route

Delete Route outbound

Edit Route

Route Name:  outbound    
Trunk Sequence:
0            SIP/sip.mydivert.com

Set Destination

 Valid for all numbers    (radio Button here is checked)

 Numbers starting by: 
     Separate prefix    

 Custom Dial Patterns:  

Options

Route Password: 
Extension:  
____________________________________________________________________________-

If you need the settings I am using at my service providers, please let me 
know.  I would be happy to add documentation to the project, but I am still new 
to sip/pbx/voip.  I am NOT a programmer or coder, just a knowledgeable user (I 
used to do desktop support for a living for 5 years).  Also, I have never 
edited a wiki before, so perhaps I can submit a guide once it is all sorted 
out?  Let me know!
Cheers!

Original comment by jerald...@gmail.com on 25 Aug 2010 at 2:17

GoogleCodeExporter commented 9 years ago
I don't understand the need for pbxes.org. Someone who starts with Sipdroid 
(which is a magnet for pbxes) would have one. But there are so many simpler and 
more direct VoIP providers... CSipSimple works with NexVortex and Callcentric 
just fine.

Or is it a way to get past the GSM providers prohibiting VoIP? I don't know 
anything about pbxes except it looks really complex to set up (as shown above! 
:-)).

Original comment by dc3de...@gmail.com on 25 Aug 2010 at 10:03

GoogleCodeExporter commented 9 years ago
I don't know much about the other services, I have meant to look, but haven't 
found the time.  It works as a voip aggregator that makes multiple voip 
accounts possible on the same account/device.

This is required if you want the cheapest services for different things, or if 
you live in the Netherlands where by law your landline/voip MUST be registered 
to an address.  My main provider 12voip is RIDICULOUSLY low priced for outgoing 
calls, but because of this stupid law here in the Netherlands, they are unable 
to offer an inbound line to me.  So after looking around, I found mydivert.com, 
and they do inbound lines in the Netherlands for 4.00 Euro a month.

It really complicates things, I agree, but it has its uses...

And yes, I started with SIPdroid after being pissed off that Skype paid 
unlimited service doesn't allow calls to mobile phones all of a sudden (Policy 
change recently, no notice or email about it, they posted it in their terms of 
service on their website).  But I like to have a choice of software...

Original comment by jerald...@gmail.com on 25 Aug 2010 at 10:31

GoogleCodeExporter commented 9 years ago
Afaik pbxes is the only pbx service which allows free registration with a 
sipclient. Furthermore sipdroid claims that using pbxes.org, instead of 
registering different sipclients on csipsimple directly, saves battery life. 
Then there are the extensions etc. which I don't use.

Basically it's good for battery consumption but I don't know if that is true 
compared to registering clients directly to csipsimple..

Original comment by alessand...@gmail.com on 25 Aug 2010 at 10:57

GoogleCodeExporter commented 9 years ago
Yes, CSipSimple works directly on some SIP providers without using Asterisk 
server.

@r3gis.3R, first of all, I am not a SIP professional.  I have experience in 
setting up a Asterisk server and makes it working.  As far as I know, h is an 
IP PBX which deals with trunks (SIP providers) and do something further for the 
incoming and outgoing calls.  May be SIP providers just provide a number or 
address for using the VoIP features.  Asterisk will take care of the others, 
such as handling the calling time, extensions, voice recording and much more.

In addition, my Linksys SPA941 cannot work without Asterisk server in my 
initial test.

I think that I will free and happy to write document for CSipSimple.  I have 
tested almost all the apps in the Market and find that CSipSimple suits my 
taste.  The interface is quiet user-friendly and easy to understand.  Please 
allow me to access wiki page if you agreed.

So far, I do not know how to capture the screen from my Nexus One, the document 
that I write will be in text only.  

Furthermore, I will be free to be a tester too.  I have 2 Nexus One, Asterisk 
server and 2 networks with or without UTM, bluetooth earpieces, and 2 GSM 
providers.

Original comment by runner...@gmail.com on 25 Aug 2010 at 11:33

GoogleCodeExporter commented 9 years ago
I've tried a ton of different settings, and nothing has worked so far.  But 
that being said, I don't really know what I am doing, I just got PBXES working 
by hacking away at it.  It eventually decided to work for me :)

I like to learn about new stuff though, so will keep trying.  If anyone here 
has suggestions based on my PBXES setup, let me know...

Original comment by jerald...@gmail.com on 25 Aug 2010 at 11:39

GoogleCodeExporter commented 9 years ago
@jeraldsib,

Your settings that listed above makes me confuse.  Would you please list the 
following out for me to study?

(1) the name of your SIP providers (trunk as below)
(2) the login name of PBXes.org
(3) what trunk for incoming route 
(4) what trunk for outgoing route
(5) your extension 

Original comment by runner...@gmail.com on 26 Aug 2010 at 12:10

GoogleCodeExporter commented 9 years ago
@alessandro & @jeralsib : ok for pbxes. Besides they are right only one account 
save battery of course. But my goal is not to provide something for a specific 
provider. I want to let user the choice. (Freedom is a matter of choice ;) ). 
Besides in many configuration pbxes.org is not really necessary. In fact, in 
France 2 out of 3 internet access provider provide to their user a free SIP 
account linked to the phone account. And so it provide an incoming and outgoing 
sip number associated to a pstn number. And in that case, we don't need 
pbxes.org... But all tutorials with sipdroid made to explain how to configure 
our accouts starts by "create a account on pbxes.org" .... while it's 
absolutely useless in our case. That's the reason why I try to learn people on 
android that pbxes.org is not an absolute need to use voip. That said, if you 
use pbxes.org for the feature they provide and that's a conscientious choice 
that's fine. My point is just to warn you about that. There is too many field 
in informatics where users doesn't take their own conscientious choice (see the 
success of the iphone...).

@jeralsib : I'll add a pbxes.org wizard today. It will make thing easier and 
take the same labels that the one presents on pbxes.org website (I'll try to 
find where are my pbxes.org credentials to test it properly ;) )

Original comment by r3gis...@gmail.com on 27 Aug 2010 at 9:16

GoogleCodeExporter commented 9 years ago
thanks r3gis.3r, I understand your comments and I agree with you. PBX services 
such as PBXes.org serve to no purpose as long as you use one line and have no 
need for extensions. I if understand you right though, as soon as you are 
permanently registered to two or more sip providers on csipsimple, grouping 
them to a PBX service and register PBX service on csipsimple would make sense 
in terms of battery consumption!
BTW I tried to find other free PBX services, but so far only PBXes.org 
registers to the sip provider for free (voxalot does this for a subscription). 
I don't like having my calls go through a german server anyway, but either put 
up with it or register my accounts directly to csipsimple and use a little more 
battery...

Original comment by alessand...@gmail.com on 27 Aug 2010 at 9:55

GoogleCodeExporter commented 9 years ago
@jeralsib : pbxes wizard added. To add a pbxes.org account now:
* Add an account 
* Expand the world wide providers
* Select Pbxes.org
* On username copy the username value of your Extension (on the website, on the 
extension section, under Device option). In front of username, just copy that.
* On password... the password (under the username on the website)
* Save
It should register.

If not your network maybe doesn't allow you to register sip.

Let me know if it helps.

Original comment by r3gis...@gmail.com on 27 Aug 2010 at 9:49

GoogleCodeExporter commented 9 years ago
OK, do I have to download the SVN or has a new version been posted to the 
marketplace?  I am traveling do I will have to stop by a McDonalds for free 
wifi and update my apps...  This wifi I have at my hotel is IP locked...  
Logging on with the phone is not possible and I forgot my USB cable...

Original comment by jerald...@gmail.com on 27 Aug 2010 at 11:21

GoogleCodeExporter commented 9 years ago
Answering my own questions here:

To install an apk that is not on the market download a file explorer for 
android like EStrongs file explorer.

Than remove your sdcard from the phone and put it into a computer somehow.

copy the APK to the sdcard and reinsert it into the phone

Use Estrongs file exporer to navigate to the APK, click it, and it will install.

Answer to question number 2, how do you set up pbxes.org:

In the new and MUCH improved setup wizard (You guys ROCK!!!!) go to new 
international account and enter the following for the account setup:

account:  me-100  (where me is your user name, and the -100 is your extension)
Password:  your password for pbxes.org.

And THAT is it...  The old guide is no longer needed, and you can close this 
issue!

Original comment by jerald...@gmail.com on 30 Aug 2010 at 7:00

GoogleCodeExporter commented 9 years ago
Oh sorry, I forgot to reply your comment! Was in a tab of my browser that I 
closed before submitting :).

I wrote a wiki page for dev installation 
http://code.google.com/p/csipsimple/wiki/HowToInstallDevVersion . But too late 
for you :) you find out the solution without my help. 

Happy to know that it works fine for you ! Marked as delivered for next release.

Original comment by r3gis...@gmail.com on 30 Aug 2010 at 7:52

GoogleCodeExporter commented 9 years ago
Can someone help me in setting up Magic Jack on CSIPSIMPLE?  When I set it up 
it says register error.  When it asks for the password is that my login 
password or is there another one?  I am a newbie here to SIP :)

Original comment by networkc...@gmail.com on 19 Oct 2010 at 2:33

GoogleCodeExporter commented 9 years ago
Are you trying to use the Magic Jack wizard or just Basic wizard to create your 
account in CSipSimple? You should try with Magic Jack wizard (since it's the 
only one that support the magic jack authentication).
Then for login and password I don't know exactly what is provided by MagicJack. 
You should maybe ask on this forum :
http://www.magicjacksupport.com/beta-testing-android-solution-csipsimple-w-built
-in-mjmd5-t9743-15.html
The guys on this forum asked me and helped me to test and implement a build in 
solution in csipsimple that allow to use magicjack authentication method.

According to what they ask me to write username is something like Exxxxx01 and 
password should be SIP password (probably something different from your user 
password...)
I guess that on the forum they'll be able to help you more than I could. But if 
that's something not linked to your credentials, ask me again (you can send me 
a private mail if you want).

Original comment by r3gis...@gmail.com on 19 Oct 2010 at 3:17