sasacoh / sipekphone Goto Github PK

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.

What steps will reproduce the problem?
1. The pjsip stack binds the wrong network interface 

Offer the option to select the appropriate network interface.

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... 

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

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?

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

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.

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. 

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

Make two outgoing calls to B and C and then click conference.

No voice between B and C.

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!

Cannot configure SIP listening port

Add edit field to settings window

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.

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!

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.

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.

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

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.

1. Make call to another party
2. Other party shouldn't see your identity!

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

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. 

What steps will reproduce the problem?
1. test

Improvment:
Many projects needs play audio file to remote endpoint.
Can you add this option to media server ?

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.

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.

Hi! 

Does Sipek support 64bit windows plateform?

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)

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?

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

If sip listening port is already taken the application crashed.

Must show error dialog and offer possibility to change listening port. 

What steps will reproduce the problem?
1. Make a call
2. Wait to other party release the call
3. The application crashes (rarely)

Silence suppression reduces bandwith usage but sometimes is not desirable - 
in example during call recording. Like echo suppression it should be 
configurable

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.

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!

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.

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.

Make clean separation between GUI, call control and voip proxy.  

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)

Order of enabled codecs doesn't affect codec priorites! Thanks Raenaet

hi
i want to know what's the difference between sipek and sipek2?
thanks

How to produce multiple SIP line like the x-lite or express talk softphone?

Hi .

Can I use command line ??

There are some links ?

Thank uou.

enrico

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.

Application crashes on many outgoing call attempts!

Exception in mainform.cs at UpdateCallRegister:
{"Collection was modified after the enumerator was instantiated."}

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

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