Originally created by: jerald...@gmail.com
Originally owned by: r3gis...@gmail.com
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...
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Originally posted by: jerald...@gmail.com
If I manage to figure it out on my own, I'll post it myself :)
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Originally posted by: runner...@gmail.com
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
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Originally posted by: jerald...@gmail.com
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"
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Originally posted by: jerald...@gmail.com
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 ;)
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Originally posted by: runner...@gmail.com
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
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Originally posted by: r3gis...@gmail.com
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.
Owner: r3gis.3R
Status: Accepted
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Originally posted by: jerald...@gmail.com
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!
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Originally posted by: dc3de...@gmail.com
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! :-)).
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Originally posted by: jerald...@gmail.com
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...
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Originally posted by: alessand...@gmail.com
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..
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Originally posted by: runner...@gmail.com
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.
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Originally posted by: jerald...@gmail.com
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...
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Originally posted by: runner...@gmail.com
@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
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Originally posted by: r3gis...@gmail.com
@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 ;) )
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Originally posted by: alessand...@gmail.com
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...
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Originally posted by: r3gis...@gmail.com
@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.
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Originally posted by: jerald...@gmail.com
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...
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Originally posted by: jerald...@gmail.com
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!
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Originally posted by: r3gis...@gmail.com
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 https://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.
Status: NextRelease
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Originally posted by: networkc...@gmail.com
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 :)
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Originally posted by: r3gis...@gmail.com
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).