sip

How to compile Linphone for iPhone ? Couldn't find libmediastreamer_base.a libmediastreamer_voip.a

▼魔方 西西 提交于 2019-12-02 03:43:59
I am trying to compile linphone source code. I've downloaded the code from Here. When I started the xcode all the libraries are missing. I have installed all ports specified in README file. I did google and got libraries but still I could not find libmediastreamer_base.a and libmediastreamer_voip.a files. Any help ? Have you build libraries? Go to linphone source folder and enter commands: $ cd submodules/build $ make all or if you don't want make your project GPL compliant enter this commands instead: $ cd submodules/build $ make all enable_gpl_third_parties=no This is all written in the

Mobicents presence server. How to register softphone?

北战南征 提交于 2019-12-02 03:10:38
I have installed the Mobicents Presence server following the guide . The server is installed but now I am not able to proceed further. I mean now how to test the presence or register the devices with XDM, PS and RLS. How do I find on which port the services are running? I am able to see register messages received to the server but on the softphones request timeouts. Is there any documentation which I am missing? Please help. The Mobicents SIP Presence Server is not supported anymore. There is an issue on GitHub currently in development about presence. Take a look at: https://github.com

How to integrate Asterisk server with external relational database, like mysql?

南楼画角 提交于 2019-12-01 23:19:49
My objectives: Client(SIP phone, i use 3CX phone) dial to asterisk server, asterisk then connect with external relational database(not located in the same place with asterisk server), and if database response something, asterisk server play a voice file(predefined .gsm file) to response client. What i already have: I have installed AsteriskNow in VirtualBox as a asterisk server, the client is using softphone to connect with Asterisk server in SIP channel. When the Client dial to asterisk server, the server can execute the dialplan. My question: If i want asterisk server establish connection to

Installer for Jitsi SIP Communicator

我们两清 提交于 2019-12-01 18:13:15
问题 I have to create installer for Jitsi SIP Communicator after I have done changes in it. I have searched on net and found some steps as given below: SOFTWARE REQUIREMENTS Cygwin: include libs "make, gcc & g++" http://cygwin.com/setup.exe MinGW: (only gcc, no g++) (use installation file .zip, taken into account that build.xml is looking for a sub-folder x86 y x64 into MinGW root folder) http://sourceforge.net/projects/mingw/files/latest/download Install in C:\MinGW\x86 copy mingw-make.exe to

Installer for Jitsi SIP Communicator

て烟熏妆下的殇ゞ 提交于 2019-12-01 18:12:47
I have to create installer for Jitsi SIP Communicator after I have done changes in it. I have searched on net and found some steps as given below: SOFTWARE REQUIREMENTS Cygwin: include libs "make, gcc & g++" http://cygwin.com/setup.exe MinGW: (only gcc, no g++) (use installation file .zip, taken into account that build.xml is looking for a sub-folder x86 y x64 into MinGW root folder) http://sourceforge.net/projects/mingw/files/latest/download Install in C:\MinGW\x86 copy mingw-make.exe to make.exe (folder C:\MinGW\x86{-}\bin) bzip2: (v1.0.5 in this case): install in C:\MinGW\bzip2 http:/

Mac 10.15 关闭SIP

若如初见. 提交于 2019-12-01 09:44:44
升级Mac后SIP开启了,根目录不能创建文件了 关闭 sip,终端输入 sudo mount -uw / 在我们开发过程中,有时候我们安装一些工具软件需要将文件拷贝到系统限制更改的文件夹中,甚至有时需要更改系统限制的文件,而这时Mac会提示系统文件不能修改之类的内容,而这时我们想要继续操作必须关闭Mac电脑的“系统完整性保护”机制(SIP) 1. 查看SIP状态 在终端中输入csrutil status,就可以看到是enabled还是disabled。 2. 关闭SIP 1 )重启MAC,按住cmd+R直到屏幕上出现苹果的标志和进度条,进入Recovery模式; 2 )在屏幕最上方的工具栏找到实用工具(左数第3个),打开终端,输入:csrutil disable; 3 )关掉终端,重启mac; 4) 重启以后可以在终端中查看状态确认。 3. 开启SIP 与关闭的步骤类似,只是在S2中输入csrutil enable即可。 来源: https://www.cnblogs.com/xidianzxm/p/11675863.html

syslog格式说明

∥☆過路亽.° 提交于 2019-12-01 07:51:55
1 告警日志 1.1.1 字段说明 字段名称 字段含义 access_time 告警时间 alarm_sip 受害ip attack_org 攻击组织 attack_sip 攻击ip attack_type 攻击类型 file_md5 文件md5 file_name 文件名 hazard_level 威胁级别 host 域名 host_md5 域名md5 ioc ioc nid nid rule_key 规则类型 serial_num 联动设备序列号 skyeye_type 原始日志类型 type 告警二级分类 super_type 告警一级分类 type_chain 告警子类标签编码 host_state 攻击结果 confidence 确信度 vuln_type 威胁名称 attack_chain 攻击链标签二级编号 super_attack_chain 攻击链标签一级编号 is_web_attack 是否web攻击 1.1.2 字典类型字段说明 hazard_level威胁级别: 1、2、3 低危 4、5 中危 6、7 高危 8、9、10 危急 1.1.3 范例 发送syslog的格式为 : (facility = local3,日志级别为:warning) 发送时间 客户端IP 日志类型 日志 2018-04-23 15:16:37|!172.17.20.159|

Replacement technology for TAPI?

瘦欲@ 提交于 2019-12-01 06:34:15
Is there a replacement technology for TAPI that supports third-party call control (3pcc)? I want to provide the following 3pcc functionalities in an application: Outgoing call: User clicks at a button in the application. The user's phone goes off hook, and the callee's phone rings. The callee's phone shows the phone number of the callee, not the phone number used for the application. When the callee picks up the phone, the connection is established. Incoming call: When user's phone rings, the caller's number and the called number are sent to the application. The application evaluates the

Python Regular Expression for SIP URI variables?

和自甴很熟 提交于 2019-12-01 06:03:52
I am using this regular expression for SIP (Session Initiation Protocol) URIs to extract the different internal variables. _syntax = re.compile('^(?P<scheme>[a-zA-Z][a-zA-Z0-9\+\-\.]*):' # scheme + '(?:(?:(?P<user>[a-zA-Z0-9\-\_\.\!\~\*\'\(\)&=\+\$,;\?\/\%]+)' # user + '(?::(?P<password>[^:@;\?]+))?)@)?' # password + '(?:(?:(?P<host>[^;\?:]*)(?::(?P<port>[\d]+))?))' # host, port + '(?:;(?P<params>[^\?]*))?' # parameters + '(?:\?(?P<headers>.*))?$') # headers m = URI._syntax.match(value) if m: self.scheme, self.user, self.password, self.host, self.port, params, headers = m.groups() I need to

Replacement technology for TAPI?

最后都变了- 提交于 2019-12-01 05:22:36
问题 Is there a replacement technology for TAPI that supports third-party call control (3pcc)? I want to provide the following 3pcc functionalities in an application: Outgoing call: User clicks at a button in the application. The user's phone goes off hook, and the callee's phone rings. The callee's phone shows the phone number of the callee, not the phone number used for the application. When the callee picks up the phone, the connection is established. Incoming call: When user's phone rings, the