Messages

1

Linux / the sip status abnormal

« on: October 10, 2014, 07:52:01 AM »

I'm using yate as our experimental tools. I've made the secondary development based the yate source code. I only modified the code in yate/engine/Socket.cpp and yate/yateclass.h( to chage the behavior of receiving and sending voice-packet).
Now I encounter a strange problem.

start the yate client, and make a phone call, every thing is ok.(the status showing in the yate client will be "ringing --> accept")
However, when i make the second phone call, the called subscriber receive the phone call, but the status showing in the yate client of the caller is still "ringing"! And after a few seconds(about 6-16 seconds), the caller will hang up automatically.

And in the following phone calls, the caller will have a very high likehood of encounting this problem.(yes, this problem is not appearing all the time).

but if I restart the yate client, and make the first phone call, everything will be ok. The following phone calls still have this problem.

Does anybody know the reasons or solutions?

2

Linux / Re: how to modify the yate client's ui

« on: June 04, 2014, 01:16:26 AM »

QComboBox selection changed (the activated() signal) is automatically signaled by QtWindow (see QtWindow::doInit() in qt4client.cpp).
The handler for it can be implemented in any ClientLogic descendent. See ClientLogic::select(), the handler for UI selection change notifications (Ref. yatecbase.h, engine/ClientLogic.cpp)

Tell me if you need more info.

Thank you very much!!!!marian!
What you said is what I need! Thanks!

3

Linux / how to modify the yate client's ui

« on: June 03, 2014, 10:04:42 PM »

I want to modify the yate client's ui, and add some new functions into the configuration.
I use qt designer to modify the yate/Share/Skins/Default/settings.ui to add a combo box into the configuration interface.
However, I also need to add a slot to do something. Anybody knows how to add a slot and connect the signal, emitting by my combo box, to the slot that I defined?

4

Linux / how to restore the audio data without udp header

« on: March 28, 2014, 12:08:50 AM »

The packets of audio data transported between 2 yate clients should be in this form: [mac header][ip header][udp header][rtpRawPacket]
I'm making an experiment, I want to transport the audio data in this form: [mac header][MPLS header][rtpRawPacket]

I find out that: the yate-client5(http://voip.null.ro/tarballs/yate5/yate-5.2.0-1.tar.gz) send out the rtp packets using udp(using the function Socket::sendTo). In order to send out a raw rtp packet encapsulated in an ether frame, I overload the sendTo function in the class Socket. I send out the frame successfully.(in clientA)

and in clientB, I overload the recvFrom function in the class Socket, and I get the Frame, de-encapsulate the raw Rtp packet, but I don't know which funtion or object should get this raw rtp packet to restore the audio data. Since my clientB cannot restore the audio data, now it works sliently.

Does anyone know how to solve this problem?

5

Linux / Re: the client cannot make sound in centos

« on: March 24, 2014, 09:51:32 PM »

Try calling locally the tonegen module: type tone/ring in call destination area and press 'Call'. You should hear a ring tone.
If you have a server between you and the party you are talking to try calling it directly.
Try using oss (most probably won't help, but try it anyway)

For the next log(s) please start yate with -Dt to see the time a message is logged.

And forget about the QT sounds not available warning, it has nothing to do with call media.

marian，thank you very much!!!!Now my yate client works perfectly!
Here is my installing steps, I hope i can help someone encounters the same problem.
I'm using ubuntu 12.04(i just point out the 2 most important dependencies: libqt4-dev and libasound2-dev, the other dependencies(for example g++, autoconf) should be installed according to the warning or error duaring your installing)
1:download the sourcecode(http://voip.null.ro/tarballs/yate5/yate-5.2.0-1.tar.gz)
2: tar -zxv -f yate-5.2.0-1.tar.gz
3: sudo apt-get install libqt4-dev #important
4: sudo apt-get install libasound2-dev #the most important step, if you are not using ubuntu(maybe centos), you can use yum search, or google(keyword: centos install alsa headers)
5: cd yate
6: ./autogen
7: make
8: sudo make install

how to start client:
9: cd client
10: sudo ./run-qt4
11: stop the ringer(setting->options->telephony, and uncheck the "Ringer")

6

Linux / Re: the client cannot make sound in centos

« on: March 23, 2014, 09:04:58 PM »

The alsa module wasn't built or is not loaded.
The default sound module is alsachan.
Install alsa devel package, run configure again and re-build yate.

You can use the osschan module. Change yate-qt4.conf:

[client]
device=oss//dev/dsp

I install the alsa module, rum configure again, re-build yate and install it again.
It works, my client can send out rtp/rtcp packets, but i still cannot hear anything and the other client still cannot hear what I say(he just hears some stable noise)......
when i'm making a phone call , the terminal warns that:
"<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA short write, writei wrote 138 frames
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred
<WARN> ALSA write underrun occurred"

and the QSound warning still exists:"<client:WARN> QT sounds are not available"

and I've uploaded the new log
do you have any idea about it, marian?

7

Linux / Re: the client cannot make sound in centos

« on: March 20, 2014, 08:58:53 PM »

Thanks for your reply，i will try it！

8

Linux / [SOLVED]the client cannot make sound in centos

« on: March 19, 2014, 04:43:26 AM »

I build and install yate5 from source code(http://voip.null.ro/tarballs/yate5/yate-5.2.0-1.tar.gz)
when i run "./run-qt4", after all modules are loaded, i get a warning showed in the terminal:"<client:WARN> QT sounds are not available"
and when a call is comming or i make a call, the gui shows:
"audio failure
Failed to open audio.
Please check your sound card"

i use wireshark to capture the packets, it shows that:
1: i can get all the sip messages correctly
2:i can receive all the rtp packets from anther yate-client(running in windows)
but, my yate-client doesn't send out any rtp-packets!! And although i can receive all rtp packets from another client, i can hear anything through my earphone.

and i also test my earphone and microphone, they work fine.

i have this method:uncheck "setting->options->telephony->Ringer", it doesn't work.

I've uploaded the log.

I hope i've made the problem clear....