jinul/sipeksdk

Automatically exported from code.google.com/p/sipeksdk

sipeksdk's Issues

SIPEK Code

What is the expected output? What do you see instead?
1) To Make a call using SIPEK.

What version of the product are you using? On what operating system?
v2.0.50727 on Windows 7

Please provide any additional information below.
I referred your code and did programming but its not working for me.
I am new to VoIP. I don't understand, how this will work?
I need Server configuration. Does any information available to do programming 
from the scratch? When I click for calling, it does not call and  when I debug, 
it shows Calling number blank.
Please let me know, how to configure server IP and port? and how to receive a 
call and how to make a call
Thanks.

Original issue reported on code.google.com by [email protected] on 23 Apr 2014 at 11:36

ADD NEW AUDIO CODEC

I WANT TO ADD NEW AUDIO CODEC TO OUR SYSTEM 

HOW CAN I DO THIS AND MANAGE IT 

PLZ ANY HELP >>>

Original issue reported on code.google.com by [email protected] on 4 Jun 2011 at 8:37

Log cannot be disabled

What steps will reproduce the problem?
1. change the logLevel = 0 in SipConfigStruct does not disable pjsip.log  
2. it seems this logLevel doens't work .
3.

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


by setting the logLevel to different value (0,1,2,3,4,5), the log file 
shall be different, 0 shall complete disabled the log)

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

Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 11 Oct 2009 at 12:35

Incoming call cannot display the caller name

What steps will reproduce the problem?
1. Incoming call cannot display the caller name
2. pjsipCallWrapper.cs  onCallIncoming
3. sturi  is in <sip:xxxxx:@[email protected]>   not display name

What is the expected output? What do you see instead?
the sturi shall be  "XXXDisplayName"<sip:xxxx@[email protected]>

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 11 Oct 2009 at 12:38

Make call doesn't work correctly

Even in Sipek Softphone, after user three-four times has made call and released 
it - dll_make_call() falls in faulted state and always returns -1,
so user can't make calls.

Original issue reported on code.google.com by [email protected] on 6 Jul 2010 at 1:31

OnUserRelease takes up to 20 seconds

I have a problem. I call a device and if i get a ringing state i directly send 
a onUserRelease to quit the call. But mostly this task takes about 20 seconds, 
and my credits get charged. Thats exactly what i didnt want.

 I just want to check for ringing state (see if device is available) and then quit.

Original issue reported on code.google.com by [email protected] on 23 Mar 2011 at 7:14

How can I call a mobile or landline phone?

How can I call a mobile or landline phone?

Original issue reported on code.google.com by [email protected] on 23 Dec 2009 at 3:19

No Sound device

What steps will reproduce the problem?

Just launch any sample app on a machine with no sound device.


The thing is. I want to do an application that just makes an test call to a 
number i specify. It is only necessary to make the call and then hang up. Its 
like a status check.

But no Microphone Device is on that machine, and in the log is an error "no 
sound device....".

Will there be any option to use the library without microphone device. 

Original issue reported on code.google.com by [email protected] on 13 Jan 2011 at 12:22

No audio packets if you are not actively speaking into the microphone

What steps will reproduce the problem?
1. Initiate a call.
2. Do not speak into the microphone
3.

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

I'm trying to inject a stream into the outgoing audio, but if you are not 
actively speaking into the microphone, no voice packets are sent.

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

Windows 7, latest build

Please provide any additional information below.

This looks like some sort of echo cancellation.  Is there a way to turn this 
off?  Or always have the sdk send rtp packets?

Original issue reported on code.google.com by [email protected] on 2 Aug 2012 at 2:34

AccessViolationException in CallManager.Initialize

AccessViolationException

bei pj_pool_release(pj_pool_t* )
   bei dll_shutdown()
   bei Sipek.Sip.pjsipStackProxy.dll_shutdown()
   bei Sipek.Sip.pjsipStackProxy.shutdown()
   bei Sipek.Sip.pjsipStackProxy.initialize()
   bei Sipek.Common.CallControl.CCallManager.Initialize(IVoipProxy proxy)
   bei Sipek.Common.CallControl.CCallManager.Initialize()

i am calling a fax device, normal phone seems to be working.

dont ask me why fax, its just a test for the availability of devices :)

Original issue reported on code.google.com by [email protected] on 2 Mar 2011 at 10:59

Issue in Conference with PJSIP, SIPEKSDK and FreeSWITCH

What steps will reproduce the problem?
1. Installed FreeSWITCH SIP server with IP 192.168.0.1 with extension 1000 to 
1019
2. Downloaded the latest pjsipDll.dll and SipekSdk.dll from trunk and developed 
a sample softphone as per the tutorial in the sipek website
3. Registration is successful for extension 1002 in my PC with IP 192.168.0.2
4. I am testing the conference by calling the first number by 
CallManager.CreateSmartOutboundCall()
5. Put the first call on hold once it's active
6. Call the second number
7. Put the calls on conference once the second call gets active

What is the expected output? What do you see instead?
In conference each party should hear voices of other 2 parties. But I am able 
hear voices of both the parties and similarly vice versa. But, Party 1 is not 
able to hear Party 2 and similarly vice versa

What version of the product are you using? On what operating system?
Latest pjsipDll.dll and SipekSdk.dll from trunk, Windows XP Professional, SP3. 
Microsoft Visual Studio 2008 for creating the softphone.

Please provide any additional information below.
FreeSWITCH is installed in Cent OS 5.2 and the conference is working fine when 
i use x-lite softphone.

Original issue reported on code.google.com by [email protected] on 8 Jul 2010 at 10:09

Detect when the remote user answers the call

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?
I am trying to charge for outgoing phone calls but I cannot determine when the 
outbound call gets answered.  Is there a way to do this or should I focus on 
spinning my own way?  Originally I thought EStateId enum would do the trick but 
that does not appear to be the case.

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

Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 29 Dec 2011 at 4:50

What if internet connection drop ...........

What steps will reproduce the problem?
1. What if internet connection drop (disconnected for isp unexpectedly)? 
2.
3.

What is the expected output? What do you see instead?
During a call, call state should change to TERMINATED or NULL. 
Call state does not change.

If not during a call, Registration State should change.
Registration State does not change.

What version of the product are you using? On what operating system?
Windows 6.1 x86.

Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 28 Sep 2010 at 4:45

regarding pjsipwrapper updation according to pjsip stack 2.0.1 (video support)

What steps will reproduce the problem?
1. use currently available pjsipwrapper source for making dll
2. after generating dll ,use that in your application.
3. it is not supporting for call but call is supporting by pjsipstack 1.10 when 
i am using this.

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

call should work with pjsip 2.0.1 also as pjsip 1.10
What version of the product are you using? On what operating system?

rightnow i am using pjsip stack 2.0.1 for video support ans using pjsipwapper 
for windows which is currently available.


Please provide any additional information below.

Original issue reported on code.google.com by [email protected] on 25 Apr 2013 at 6:56

Codec

Hi.

I want to use codec G729 with your application. I am wondering for this issue.
Please guide me how to use it with your windows mobile dialer example.?

Thanks 
Adeel 

Original issue reported on code.google.com by [email protected] on 16 Apr 2012 at 11:17

video support

Hi,

It will be helpful if sipek SDK is adorned with the video support as PJSIP 2.0 
does.

Please update the source code.

Thanks.

Original issue reported on code.google.com by [email protected] on 31 May 2014 at 6:04

Can't transfer a call

What steps will reproduce the problem?

Download and test the simple phone program,
add a text box (txTransfer) and a button (btTransfer)
dial a call
hit the button with this underlying code

private void btTransfer_Click(object sender, EventArgs e) {
   CallManager.onUserTransfer(_call.Session, txTransfer.Text);
}

Nothing happens..

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

The call should transfer to the extension typed into the txTransfer text box

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

Lastest dlls as from the site (v3 ??)

Please provide any additional information below.

None I can think of!

Original issue reported on code.google.com by [email protected] on 12 Mar 2012 at 5:13

Attachments:

• Form1.cs

Cannot open SipekSdk_mobile.cproj and convert it

Hello,

I'm using VS 2008, whenever i try and open the SipekSdk_mobile.cproj file it 
shows me the window for upgrading and then fails, so i can't open it and make 
the sln file to be able to rebuild

any ideas?

Original issue reported on code.google.com by [email protected] on 26 Oct 2010 at 11:39