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Yet Another Telephony Engine - UNOFFICIAL mirror

Home Page: http://yate.null.ro/

License: GNU General Public License v2.0

Shell 0.20% C++ 74.63% Objective-C++ 0.05% C 21.67% Objective-C 0.07% Perl 0.23% PHP 1.12% JavaScript 0.11% Python 0.70% Pike 0.11% CSS 0.07% Batchfile 0.01% Inno Setup 0.16% Makefile 0.50% M4 0.38%

yate's Introduction

		YATE - Yet Another Telephony Engine
		-----------------------------------

  The YATE project aims to be a fully featured software PBX.

  It was created to alow developers and users to have more functionality and
  scalability. To reach this goal YATE is built from two kinds of components:
    1. The main engine - telengine.
    2. Modules - routing modules
               - drivers
               - script language bindings
	       - billing modules
  
  Its license is GPL with exceptions (in case of linking with proprietary
  software). We have chosen this license to help the growth of this project.


Building YATE Software
----------------------

YATE have been tested on Linux and Windows and was ported to FreeBSD.
Please report bugs at [email protected]

To build Yate the quick way is:
  ./configure
  make
  make install

If you are lacking both doxygen and kdoc you will need to install without the
API documentation:
  make install-noapi

1. Building the engine

You have just to run 'make engine' in the main directory.

2. Building the modules.

Run 'make modules' in the main directory or 'make' in the modules directory.

3. Building the test modules.

Run 'make test' in the main directory or 'make' in the test directory.

After you have create the test modules use 'mktestlinks' in the modules
directory to make links from test modules into modules directory.

4. Building the classes API documentation

Run 'make apidocs' in the main directory. You will need to have kdoc or
doxygen installed.


Alternatively you can just 'make everything' in the main directory which will
build them all.


Running YATE
------------

You can run YATE directly from the build directory - just use 'run' script
from the main directory.
  ./run -vvv

You can also install YATE - then you can run it from anywhere.
  yate -vvv

On the command line you can use '-v' to increase the verbosity level. If in
doubt run "run --help" (or "yate --help" if installed) to get a list of
possible options. There is also a manual page - "man yate" and read.

While running the following signals and keys are trapped and used:
  - SIGTERM and SIGINT (Ctrl-C) will cleanly stop the engine
  - SIGHUP and SIGQUIT (Ctrl-\) will reinitialize the modules
  - SIGUSR1 will gracefully restart a supervised engine
  - SIGUSR2 will forcefully restart a supervised engine


Configuring YATE
----------------

Some samples for the configuraton files can be found in the conf.d directory.
Note that you must rename them without the .sample extension or create symlinks
to them.

yate's People

Contributors

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yate's Issues

client call auto released after 15 seconds

Hi!
My Sip phone solution is:
Sip server:Yate 6.1.0,windows 2012
Sip client A:linphone desktop,linphone web plugin
Sip client B:linphone linux
Sip server with two IP,Public network static IP:222.178.179.xxx,local ipv4:10.0.21.x
Sip client register to 222.178.179.xxx,client A and client B registed all ok.
my issue is:when i call client B from client A,after 15 seconds,the call released by client A,I notice that client A will keep alive to 10.0.21.x and send message to 10.0.21.x witch can't reach, 10.0.21.x is local IP,I need to hide local IP,only open public Ip 222.178.179.xxx to all client,I try to ysipchan.conf,but failed.
can anyone help me,please,thank you.

Yate can't connect to SIP provider

REGISTER sip:permngn.usi.ru:5060 SIP/2.0
Contact: <sip:[email protected]:5160>
Expires: 600
To: <sip:[email protected]:5060>
Via: SIP/2.0/UDP 192.168.10.97:5160;rport;branch=z9hG4bK814382474
From: <sip:[email protected]:5060>;tag=533209016
Call-ID: [email protected]:5060
CSeq: 127 REGISTER
User-Agent: YATE/4.1.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

SIP/2.0 100 Trying
From: <sip:[email protected]:5060>;tag=533209016
To: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 127 REGISTER
Via: SIP/2.0/UDP 192.168.10.97:5160;rport=5160;branch=z9hG4bK814382474
Content-Length: 0

SIP/2.0 407 Proxy Authentication Required
From: <sip:[email protected]:5060>;tag=533209016
To: <sip:[email protected]:5060>;tag=1824228324
Call-ID: [email protected]:5060
CSeq: 127 REGISTER
Via: SIP/2.0/UDP 192.168.10.97:5160;rport=5160;branch=z9hG4bK814382474
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTQzOTg4MTk5Mjk2MGZkYTBiMzRmM2E1M2JhZmU0YjhhYWY3MjAwZDhmM2Fh",stale=false,algorithm=MD5,qop="auth,auth-int"
Content-Length: 0

REGISTER sip:permngn.usi.ru:5060 SIP/2.0
Contact: <sip:[email protected]:5160>
Expires: 600
To: <sip:[email protected]:5060>
Via: SIP/2.0/UDP 192.168.10.97:5160;rport;branch=z9hG4bK1260412158
From: <sip:[email protected]:5060>;tag=533209016
Call-ID: [email protected]:5060
User-Agent: YATE/4.1.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
CSeq: 128 REGISTER
Content-Length: 0

SIP/2.0 100 Trying
From: <sip:[email protected]:5060>;tag=533209016
To: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 128 REGISTER
Via: SIP/2.0/UDP 192.168.10.97:5160;rport=5160;branch=z9hG4bK1260412158
Content-Length: 0

SIP/2.0 407 Proxy Authentication Required
From: <sip:[email protected]:5060>;tag=533209016
To: <sip:[email protected]:5060>;tag=368007154
Call-ID: [email protected]:5060
CSeq: 128 REGISTER
Via: SIP/2.0/UDP 192.168.10.97:5160;rport=5160;branch=z9hG4bK1260412158
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTQzOTg4MTk5Mjk2MGZkYTBiMzRmM2E1M2JhZmU0YjhhYWY3MjAwZDhmM2Fh",stale=false,algorithm=MD5,qop="auth,auth-int"
Content-Length: 0

\

Invalid signature

Hello,

I have added key:

~# wget -O - http://debian.ctm.ru/logs/repository_key.asc | apt-key add -
--2017-12-15 15:01:20--  http://debian.ctm.ru/logs/repository_key.asc
Resolving debian.ctm.ru (debian.ctm.ru)... 91.220.57.8
Connecting to debian.ctm.ru (debian.ctm.ru)|91.220.57.8|:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 1657 (1.6K) [application/octet-stream]
Saving to: ‘STDOUT’

-                                             100%[================================================================================================>]   1.62K  --.-KB/s    in 0s      

2017-12-15 15:01:20 (308 MB/s) - written to stdout [1657/1657]

OK

Then I've gotten error on apt-get update:

Err:4 http://debian.ctm.ru/vir stable InRelease                                                                                                                                       
  The following signatures were invalid: 4F43FB327ADB1B8D1BB605B9EED98229F072B85E

Please let me know how to fix

Yate support for m3ua

Hi Team,

Please let us know does yate supports m3ua ? if does please let us know how we can configure.

Chees!
Vinod Dharashive

SS7 ISUP not working on Debian vir packages

Я опубликовал этот баг 417 в Mantis yate http://yate.null.ro/mantis/view.php?id=417

Я обнаружил, что версия Yate в vir Debian репозитории не декодирует SS7 ISUP части SIP пакетов. Проверил другие операционки - бинарники yate для Windows декодируют SS7 ISUP. Собрал на Debian 8 yate из исходников deb-src vir repo (debuild -us -uc) - не декодирует ISUP. Собрал на Debian 8 yate руками из исходников svn - SS7 ISUP декодирует. Более того, поставил при помощи alien rpm пакет yate от CentOS - декодирование SS7 ISUP работает.

Похоже, что-то не то в ветке vir? Я пробовал разобраться, но так быстро не получилось - diff великоват (308K).

Докментация

Я бы хотел использовать Yate как Wbesocket SIP сервер. Но я не могу даже настроить sip listener.
Может быть ветка sip-ws вообще не юзабельна в данный момент? Если так, то я хотел бы помочь, дописав необходимое

Queues handling problem in queue_out.php

I've found there is a bug (probably) at this moment in queue_out.php script. This bug prevents correct call distribution between operators in a queue. I've made patch fixing this problem, so I would like to post it here.

Lazyrec Module

Hello! Please could I ask for some help ?

Is it necessary to install lazyrec.cpp ? Or is this problem now irrelevant?

"Current wavechannel implements recording in voice data handling thread via blocking write. If any backlog builds in the operating system voice quality decreases. Solution implemented in attached file is to buffer data and write it in separate thread."

I found reference to it here also : http://yate.null.ro/mantis/view.php?id=68

but unfortunately I've not be able to find any instructions.

If required, please could you point me in the right direction?

Thank you!
Regards
Jamie

Append .log to log file name (debian packaging)

<Centurion_Dan> virrus_, Logs don't need to be in a subdirectory (unless separating out logs for different parts of yate). But use /var/log/yate.log as opposed to /var/log/yate - not using the .log extension can break logrotate if it's set to use date extensions.

Multifon SIP provider support

I noticed wiki page on Multifon SIP provider, so I thought yate supports it. Alas, it does not seem to work for me =(
.
.
.
Debug log (personal data removed, I believe :D):

<sip:INFO> Sending 'REGISTER sip:multifon.ru' 04706468 to 193.201.229.35:5060
------
REGISTER sip:multifon.ru SIP/2.0
Contact: <sip:[email protected]:5060>
Expires: 60
To: <sip:[email protected]>
Call-ID: [email protected]
Via: SIP/2.0/UDP 93.92.200.251:5060;rport;branch=z9hG4bK314370635
From: <sip:[email protected]>;tag=1893154680
CSeq: 49 REGISTER
User-Agent: YATE/3.3.2
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
Content-Length: 0

------
<sip:INFO> Received 487 bytes SIP message from 193.201.229.35:5060
------
SIP/2.0 407 Proxy Authentication Required
To: <sip:[email protected]>;tag=475232463135364141E1B100
Call-ID: [email protected]
Via: SIP/2.0/UDP 93.92.200.251:5060;received=93.92.200.251;branch=z9hG4bK314370635;rport=5060
From: <sip:[email protected]>;tag=1893154680
CSeq: 49 REGISTER
Proxy-Authenticate: Digest nonce="MTIzNDU2Nzg5MDpyYW5kb21zdHJpbmc",opaque="MTIzNDU2Nzg5MDpyYW5kb21zdHJpbmc",algorithm=md5,realm="BREDBAND",qop="auth"
Content-Length: 0

------
<sip:INFO> Sending 'REGISTER sip:multifon.ru' 047062B8 to 193.201.229.35:5060
------
REGISTER sip:multifon.ru SIP/2.0
Contact: <sip:[email protected]:5060>
Expires: 60
To: <sip:[email protected]>
Call-ID: [email protected]
Via: SIP/2.0/UDP 93.92.200.251:5060;rport;branch=z9hG4bK1823350104
From: <sip:[email protected]>;tag=1893154680
User-Agent: YATE/3.3.2
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
CSeq: 50 REGISTER
Proxy-Authorization: Digest username="79312345678", realm="BREDBAND", nonce="MTIzNDU2Nzg5MDpyYW5kb21zdHJpbmc", uri="sip:multifon.ru", response="a0ba7111a822428b9fd080be7ab19520", algorithm=MD5, opaque="MTIzNDU2Nzg5MDpyYW5kb21zdHJpbmc"
Content-Length: 0

------
<sip:INFO> Received 388 bytes SIP message from 193.201.229.35:5060
------
SIP/2.0 500 Internal Server Error
To: <sip:[email protected]>;tag=C7F432463135364146E1B100
Call-ID: [email protected]
Via: SIP/2.0/UDP 93.92.200.251:5060;received=93.92.200.251;branch=z9hG4bK1823350104;rport=5060
From: <sip:[email protected]>;tag=1893154680
CSeq: 50 REGISTER
Contact: sip:[email protected]:5060;expires=180
Content-Length: 0

------
<sip:WARN> SIP line 'sip:[email protected]' logon failure 500: Internal Server Error
Unregistered account sip:[email protected] reason: Internal Server Error

.
.
.
client_accounts.conf:

[sip:[email protected]]
enabled=true
protocol=sip
username=79312345678
savepassword=true
server=multifon.ru
domain=multifon.ru
outbound=sbc.megafon.ru
options=
resource=
port=
interval=60
authname=79312345678
datadirectory=956961082_4030470697_3831846117
password=pass

.
.
.
956961082_4030470697_3831846117\account.conf:

[general]
account=sip:[email protected]
protocol=sip
username=79312345678
domain=multifon.ru
server=multifon.ru

Deb package build failed

dpkg-buildpackage failed.

dpkg-buildpackage: source package yate
dpkg-buildpackage: source version 5.5.1~vir2016070901
dpkg-buildpackage: source distribution UNRELEASED
dpkg-buildpackage: source changed by Vasily i. Redkin <[email protected]>
dpkg-buildpackage: host architecture amd64
 dpkg-source -Zgzip -z1 -Zgzip -z1 --before-build yate-5.5.1~vir2016070901
 fakeroot debian/rules clean
./autogen.sh
make: ./autogen.sh: Command not found
debian/rules:20: recipe for target 'configure' failed

Ошибка генерации SIP-поля CSeq

Поймал очень противную багу в последней версии ята 5.5.0
Суть в том, что после второго инвайта счетчик CSeq не сохраняется в нужном месте внутри движка ята. Как итог следующая команда после этого инвайта будет иметь ТАКОЙ же номер CSeq как и второй инвайт. Что является ошибкой и один провайдер у меня из-за этого глючил. Все последующие команды уже имеют корректный инкрементированный номер.

Исходящий вызов:

Запрос INVITE
CSeq: 11 INVITE

Ответ 401 Unauthorized
CSeq: 11 INVITE

Запрос INVITE
CSeq: 12 INVITE

Ответ 100 Trying
CSeq: 12 INVITE

Запрос INFO
----> Ошибка, здесь должно быть 13!!<-----
CSeq: 12 INVITE

Запрос INFO
CSeq: 13 INVITE

Запрос INFO
CSeq: 14 INVITE

Или еще неккоретный пример:

Запрос INVITE
CSeq: 100 INVITE

Ответ 401 Unauthorized
CSeq: 100 INVITE

Запрос INVITE
CSeq: 101 INVITE

Ответ 100 Trying
CSeq: 101 INVITE

Запрос BYE
----> Ошибка, здесь должно быть 102!!<-----
CSeq: 101 INVITE

А вот как должно быть:

Запрос INVITE
CSeq: 122 INVITE

Ответ 401 Unauthorized
CSeq: 122 INVITE

Запрос INVITE
CSeq: 123 INVITE

Ответ 100 Trying
CSeq: 123 INVITE

Запрос BYE
CSeq: 124 INVITE

Вот что говорит rfc 3261
https://tools.ietf.org/html/rfc3261
"
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request within a dialog and
is a traditional sequence number.
"

Честно говоря мне не хватило скила корректно поправить ошибку, местный код слишком запутанный. Я почти уверен, что вы сталкивались с этой проблемой. Может быть уже есть патч?

Hardcoded icon path in launcher for GNU/Linux systems

Hi,

currently the icon path specified in the created launcher file is set to:
Icon=null_team-48.png

(see https://github.com/vir/yate/blob/456b507c85503d7de65fd33328f149ce41fffa9c/clients/yate-qt4.desktop)

This prevents icon theming without editing the launcher file.

I'd like to ask you to change the icon location as well as the icon path line in the launcher according to the freedesktop.org standards.

Please find more info here:
http://standards.freedesktop.org/icon-theme-spec/icon-theme-spec-latest.html
https://github.com/Foggalong/hardcode-fixer/wiki/What,-Why-&-How


According to the above resources standard icon locations are:

/usr/share/icons/hicolor/[size]/apps/[icon name]
~/.local/share/icons/hicolor/[size]/apps/[icon name]
/usr/share/pixmaps/[icon name]

Your icon multibootusb.png is already stored in a standard location (/usr/share/pixmaps).

But the standard icon line in the .desktop launcher is:
Icon=[icon name] (neither full path, nor the file extension are needed for icons in standard location)

for instance
Icon=yate
(for an icon yate.svg or yate.png stored in standard location)

Could you please adjust the icon path in the .desktop file accordingly?

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