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