sasacoh / sipekphone Goto Github PK
View Code? Open in Web Editor NEWAutomatically exported from code.google.com/p/sipek2
Automatically exported from code.google.com/p/sipek2
Sipek project! See http://sipekphone.googlepages.com/! Q: How to use sipek TLS with openser? A: First put certificate and private key created by openser to application folder and rename them to server.crt and pkey.key! Check option TLS in Sipek->Settings. Q: What to do if something goes wrong? A: Check configuration, network status, or see the sipek discussions for similar problems. Finaly analize pjsip.log file by yourself or send question to sipek discussion group. Good luck! Sasa
What steps will reproduce the problem?
1. test
Original issue reported on code.google.com by [email protected]
on 29 Oct 2008 at 8:15
What steps will reproduce the problem?
1. Make an outgoing call (asterisk)
What is the expected output? What do you see instead?
The server annoucement should be heard but there's generated alerting tone
mixed together.
Original issue reported on code.google.com by [email protected]
on 9 Mar 2009 at 12:34
Order of enabled codecs doesn't affect codec priorites! Thanks Raenaet
Original issue reported on code.google.com by [email protected]
on 11 Aug 2008 at 7:08
Hello.
How can I call a mobile or landline phone with computer using a program
Sipek Softphone?
I tried different ways, but I do not get.
For example using the program 3CX everything works, I dial: 00ZYYYXXXXXXX,
but Sipek Softphone is not working.
Original issue reported on code.google.com by [email protected]
on 23 Dec 2009 at 3:17
What steps will reproduce the problem?
1. Answer an call, and try to transfer it to a 3thd party and first
announce the caller to the 3thd party before transfering
2.
3.
What is the expected output? What do you see instead?
# Only blind transfer is supported
What version of the product are you using? On what operating system?
# Latest from svn
Please provide any additional information below.
# Answer an call, and try to transfer it to a 3thd party and first
announce the caller to the 3thd party before transfering
p.s. Is it possible to explain the conferance option in the code, I'm not
able to get this done.
Original issue reported on code.google.com by robin%[email protected]
on 17 Dec 2008 at 4:07
What steps will reproduce the problem?
1. I use de SIPEK to dial
2. But, i need do call and when call was answered play a mp3 message
3. When mp3 message finish do play de call need do hangup
Original issue reported on code.google.com by [email protected]
on 21 Oct 2009 at 3:29
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 18 Mar 2010 at 9:41
Hi .
Can I use command line ??
There are some links ?
Thank uou.
enrico
Original issue reported on code.google.com by [email protected]
on 11 Mar 2010 at 5:20
What steps will reproduce the problem?
1. A (thats me) make call to B
2. B answers
3. B transfer call to C
4. A received REFER and reply 202 Accepted (initiates new call to C)
5. A receives BYE from B and GUI remove session
The problem is that new session is not shown on GUI and the old one is
released. So GUI call list is empty and the voice connection is established!
Original issue reported on code.google.com by [email protected]
on 1 Aug 2008 at 11:45
How to produce multiple SIP line like the x-lite or express talk softphone?
Original issue reported on code.google.com by [email protected]
on 17 Oct 2009 at 3:30
hi
i want to know what's the difference between sipek and sipek2?
thanks
Original issue reported on code.google.com by [email protected]
on 9 Dec 2012 at 11:01
Hi
I am able to compile sipek2 on windows succesfully.
Does it support zrtp currently ? As Pjsip supports ZRTP implementation with a
patch from Zorg, how to add this zrtp support to sipek2 phone.
Need some help.
Original issue reported on code.google.com by [email protected]
on 6 May 2011 at 7:40
What steps will reproduce the problem?
1. connected to voipswitch
What is the expected output? What do you see instead?
No echo should be there, I'm getting echo on phone
Original issue reported on code.google.com by [email protected]
on 14 Aug 2010 at 4:21
What steps will reproduce the problem?
1. I used the SipekSDK's method setSoundDevice to change the sound device.
2. But it does not make any change to sound device.
3.
What is the expected output? What do you see instead?
Sound device must be changed. Am I missing something?
Is there any need for addition settings.
What version of the product are you using? On what operating system?
I am using WindowXP.
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 28 Jun 2011 at 6:14
What steps will reproduce the problem?
1. build on VS C# 2008
2. run application
3. check the codec list
Some strange characters appear in list.
The problem is in passing strings from unmanaged to managed code.
Original issue reported on code.google.com by [email protected]
on 25 Feb 2008 at 8:35
What steps will reproduce the problem?
1. Trying to register on a different network than the Sip server
2.
3.
What is the expected output? What do you see instead?
Expecting a registration 200
Receiving a 408 Timeout.
What version of the product are you using? On what operating system?
Most Current issue of the product, running Windows XP Service pack 3.
Please provide any additional information below.
I am able to successfully register with X-lite and other sip phones on the
client computer but not able to register with Sipek. Are there additional
settings to traverse firewalls or other outside network connectivity issues?
Original issue reported on code.google.com by [email protected]
on 23 Sep 2011 at 3:13
What steps will reproduce the problem?
1. Received calls from other softphone (such as X-lite) without any problem
2. Can't make outgoing calls
- Type number into ComboDial
- Then I press button call, but nothing happened
What version of the product are you using? On what operating system?
Windows 7 64-bit, VS2010 Express Edition
Note: When I use pjsipDll.dll r107, I can make outgoing calls to X-lite
SoftPhone, but I can't see any incoming call.
Plz help me!
Original issue reported on code.google.com by [email protected]
on 24 Dec 2012 at 7:25
What steps will reproduce the problem?
1. Select Audio in Settings.
2. Change Audio input/output to another soundcard.
3. Press Apply or OK.
What is the expected output? What do you see instead?
Change Audio input/output to the selected settings. Instead to goes back to
default.
What version of the product are you using? On what operating system?
0.3.136.108, Windows XP
Original issue reported on code.google.com by [email protected]
on 22 Jan 2010 at 1:54
What steps will reproduce the problem?
1. I built the PJSIP library 0.9.0
2. Included the pjsipdll vcproj as suggested in
https://sites.google.com/site/sipekvoip/Home/documentation/pjsipwrapper/pjsipwra
pper-for-windows
And built the project successfully to obtain pjsipDlld.dll file
3. Now I've made a cpp project where I've included the pjsipDll.h and added the
pjsipDll.dll file. In the main, I am trying to call
dll_makeCall(1234,"abcd");
I built the project then,
I receive the following error
Error 1 fatal error LNK1302: only support linking safe .netmodules; unable to
link ijw/native
.netmodule g:\July\9.0\pjproject-0.9.0\newProject\newProject\pjsipDll.dll 1 newP
roject
Is my approach correct, what do need to change to use the functions and build
my custom program for it.
I'm using SIPEK and not SIPEK2
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 29 Jun 2012 at 7:06
Attachments:
What steps will reproduce the problem?
1. in buddy list add/remove buddies
What is the expected output? What do you see instead?
New buddy in list or removed buddy from list
Original issue reported on code.google.com by [email protected]
on 8 Mar 2008 at 10:30
What steps will reproduce the problem?
1. Changing the playback/recording sound device in sipek GUI
What is the expected output? What do you see instead?
The sip stack audio should use selected device. But it doesn't change.
Original issue reported on code.google.com by [email protected]
on 22 Sep 2008 at 7:00
System.NullReferenceException: Object reference not set to an instance of
an object.
at Sipek.MainForm.LoadAudioValues()
at Sipek.MainForm.MainForm_Load(Object sender, EventArgs e)
Original issue reported on code.google.com by [email protected]
on 1 Sep 2009 at 8:50
If sip listening port is already taken the application crashed.
Must show error dialog and offer possibility to change listening port.
Original issue reported on code.google.com by [email protected]
on 6 Mar 2008 at 9:00
Improvment:
Many projects needs play audio file to remote endpoint.
Can you add this option to media server ?
Original issue reported on code.google.com by [email protected]
on 10 May 2009 at 9:49
Make two outgoing calls to B and C and then click conference.
No voice between B and C.
Original issue reported on code.google.com by [email protected]
on 19 Nov 2008 at 12:56
Hello.
I can not understand why sipek not connect to the provider voipbuster.com!
I tried to connect to this provider (voipbuster.com) through other
sip-clients, all work, but through sipek fails.
Please help, what could be the reason?
Maybe it's because of the lack of the necessary codecs?
Original issue reported on code.google.com by [email protected]
on 2 Feb 2010 at 5:35
Silence suppression reduces bandwith usage but sometimes is not desirable -
in example during call recording. Like echo suppression it should be
configurable
Original issue reported on code.google.com by [email protected]
on 29 Nov 2008 at 5:24
1. Make call to another party
2. Other party shouldn't see your identity!
Original issue reported on code.google.com by [email protected]
on 26 Sep 2008 at 1:44
Wrong account registration status. If you do not set account settings in
right order (1,2,3,4,5) the registration status will be displayed in the
wrong line.
Problem with GUI account id and sip stack account id (dynamically assigned).
The account id mapping should be implemented.
Original issue reported on code.google.com by [email protected]
on 18 Jul 2008 at 6:27
Make clean separation between GUI, call control and voip proxy.
Original issue reported on code.google.com by [email protected]
on 25 Feb 2008 at 8:50
What steps will reproduce the problem?
1. Binary application didn't work
2. Downloaded source and opened in VS 2008, converted project files to new
format
3. Right clicked GUI, rebuild
4. Running app in debug crashes.
The following line is where it crashes:
private IRegistrar _registrar = pjsipRegistrar.Instance;
in file
SipekFactory.cs
With the following error:
An attempt was made to load a program with an incorrect format.
(Exception from HRESULT: 0x8007000B)
I'm running on Windows Vista Enterprise SP1 x64 with all updates applied.
VS 2008 is my dev environment. Any idea what the problem is... as stated,
the binaries from the installer also crash before anything is displayed.
thank you!
Original issue reported on code.google.com by [email protected]
on 10 Dec 2008 at 9:17
Application crashes on many outgoing call attempts!
Exception in mainform.cs at UpdateCallRegister:
{"Collection was modified after the enumerator was instantiated."}
Original issue reported on code.google.com by [email protected]
on 6 Mar 2008 at 12:10
hi
I've used SipekSDK in my c# application
problem is that some times my application hangs
i create a dump file using task manager and opened it by windbg
here is the error
please help me
thanks
.................................................
eax=00520298 ebx=00000000 ecx=7ffde000 edx=7ffdf000 esi=0049cab8 edi=001202ee
eip=776170b4 esp=002acb88 ebp=002acbe8 iopl=0 nv up ei pl nz na pe nc
cs=001b ss=0023 ds=0023 es=0023 fs=003b gs=0000 efl=00200206
ntdll!KiFastSystemCallRet:
776170b4 c3 ret
Original issue reported on code.google.com by [email protected]
on 4 May 2013 at 11:25
Hi!
Does Sipek support 64bit windows plateform?
Original issue reported on code.google.com by [email protected]
on 9 Jan 2010 at 4:17
What steps will reproduce the problem?
1. install freeswitch (default config is ok)
2. connect gui with extension 1000
3. connect x-lite with extension 1001
4. dial 1001 from gui and answer from x-lite
5. press *2 on gui's keypad
What is the expected output? What do you see instead?
open call recording (under freeswitch/recordings/) and open it (with vlc or
what else). Recorded audio should contain leg-a in the left channel and
leg-b in the right one.
Notice that, instead, leg-a (on the left channel) is just white noise,
while leg-b is correctly recorded. Inverting gui and x-lite registration is
ok
What version of the product are you using? On what operating system?
last version (as of today) of both sipek2 and freeswitch
Original issue reported on code.google.com by [email protected]
on 25 Nov 2008 at 10:39
When called client immediately answered call with 200 ok (without 180
ringing) no transition is defined for call state machine.
The call line remains in calling state
Original issue reported on code.google.com by [email protected]
on 6 Mar 2008 at 8:57
With an incoming call the Caller Name (Calling Name) appears to be blank.
it should be available from 'CallingName' :
Dictionary<int, IStateMachine> callList =
SipekResources.CallManager.CallList;
foreach (KeyValuePair<int, IStateMachine> kvp in callList) {
string number = kvp.Value.CallingNumber;
string callername = kvp.Value.CallingName;
With SIP the full string should be in the following format:
"MyCallerID"<[email protected]>
I have looked around and the calling name does not seem to be picked up
anywhere - the number on the other hand works fine. It might be a good
idea just to expose the full raw incoming string and then give the ability
to parse it in the code...
Original issue reported on code.google.com by [email protected]
on 27 May 2009 at 4:21
Hello,
I use a DSL model with built-in SIP proxy server (Gigaset SX763 WLAN DSL
from Siemens AG). It uses a banlk in the user name and also a star in the
display name. e.g. Username is "Phone 5" and display name is "*5".
What steps will reproduce the problem?
1. If the user name field contains a space, I will see the status
messahe "Trying...".
2. If there is no bank space in the user name and id the display name
start with a star character, I get the status message "404".
What is the expected output? What do you see instead?
-Please support these two special cases or introduce some escape sequence
to enter such values in those fields.
What version of the product are you using? On what operating system?
- I used V 0.3.136.108 on Windows XP
Please provide any additional information below.
- none
Regards,
(Byju)
Original issue reported on code.google.com by [email protected]
on 8 Mar 2009 at 12:04
What steps will reproduce the problem?
1. The pjsip stack binds the wrong network interface
Offer the option to select the appropriate network interface.
Original issue reported on code.google.com by [email protected]
on 11 Mar 2009 at 7:13
What steps will reproduce the problem?
1. Make a call
2. Wait to other party release the call
3. The application crashes (rarely)
Original issue reported on code.google.com by [email protected]
on 25 Nov 2008 at 11:35
What steps will reproduce the problem?
1. install freeswitch (default configuration is ok)
2. connect as extension 1000 with gui
3. connect as extension 1001 with x-lite or another sip phone
4. call extension 1000 from x-lite
What is the expected output? What do you see instead?
expected: gui displaying call from 1000
instead: gui displays an incoming call from 'mod_sofia' (the sip stack used
by freeswitch)
What version of the product are you using? On what operating system?
last sipek binary release (as of today)
last freeswitch binary windows (.msi) distribution
Please provide any additional information below.
call extension 1001 from gui - x-lite displays an incoming call from 1000
Original issue reported on code.google.com by [email protected]
on 25 Nov 2008 at 2:48
What steps will reproduce the problem?
1. Make a call and release it
2. Change one of the configuration parameters
3. Press ok
In the background a protocol stack is restarted (shutdown, start). This
cause application crash.
Original issue reported on code.google.com by [email protected]
on 13 Sep 2007 at 6:29
Cannot configure SIP listening port
Add edit field to settings window
Original issue reported on code.google.com by [email protected]
on 6 Mar 2008 at 8:54
What steps will reproduce the problem?
1. Have a larger call log in my experience (2.5MB+)
What is the expected output? What do you see instead?
For the program to function normally. I see the program go to a not
responding status in the task manager and then eventually crash.
What version of the product are you using? On what operating system?
0.3.136.108 - Windows XP SP3 / Windows Vista Business Edition
Please provide any additional information below.
Replacing the call log with the original fixes the issue. Possible fixes:
- Parse only a portion of the call log
- Move over to something like SQLite
- Provide a GUI option to not log calls at all / clear the log
- Provide utility to rotate logs
Thank you.
Original issue reported on code.google.com by [email protected]
on 14 Aug 2009 at 3:49
What steps will reproduce the problem?
1. After installed Sipek softphone
2. Double click on the sipek softphone to launch
3. Initialize Error occur, "Init SIP stack problem! Please, check
configuration and start again! Status code 130048"
What version of the product are you using? On what operating system?
SipekSoftphone_0.3.136.108.msi
Windows XP pro
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 2 Feb 2010 at 10:16
What steps will reproduce the problem?
1. My machine has two Audio-In ( Microphone In ) ports and I intend to send a
mix of both the Inputs over the call.
What is the expected output? What do you see instead?
The person receiving the call will hear a mix of the two Audio Inputs I have.
What version of the product are you using? On what operating system?
Sipek Softphone Latest Release ( Version 3) Windows 7 64bit
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 26 Jan 2014 at 7:48
A declarative, efficient, and flexible JavaScript library for building user interfaces.
🖖 Vue.js is a progressive, incrementally-adoptable JavaScript framework for building UI on the web.
TypeScript is a superset of JavaScript that compiles to clean JavaScript output.
An Open Source Machine Learning Framework for Everyone
The Web framework for perfectionists with deadlines.
A PHP framework for web artisans
Bring data to life with SVG, Canvas and HTML. 📊📈🎉
JavaScript (JS) is a lightweight interpreted programming language with first-class functions.
Some thing interesting about web. New door for the world.
A server is a program made to process requests and deliver data to clients.
Machine learning is a way of modeling and interpreting data that allows a piece of software to respond intelligently.
Some thing interesting about visualization, use data art
Some thing interesting about game, make everyone happy.
We are working to build community through open source technology. NB: members must have two-factor auth.
Open source projects and samples from Microsoft.
Google ❤️ Open Source for everyone.
Alibaba Open Source for everyone
Data-Driven Documents codes.
China tencent open source team.