kamailio

PSTN to OpenSIPS to next SIP destination

不问归期 提交于 2021-02-10 16:44:10
问题 I have worked with Asterisk for years but I am very new to OpenSIPS. What I need is to have calls come in from our DID provider to the OpenSIPS server then redirect them to another SIP URI. Something like this: DID Origination Provider -> OpenSIPS -> next SIP server Basically I need the OpenSIPS server to sit between my DID provider and and Plivo which is basically a Twilio type service. I have installed OpenSIPS and the control panel GUI. Using the GUI I have successfully setup calls to go

How to get call info from Kamailio

女生的网名这么多〃 提交于 2019-12-11 19:43:56
问题 I have setup a Kamailio server and am able to establish calls. I need a way to get call related information like from, to, duration,etc. I have enabled the dialog module in the config but no avail. I am not well versed with config files and I am not sure if I am doing something wrong in the config file. 回答1: You need to Modify the config file to log the call related information in kamailio database tables.Here's the link You have to uncomment the lines in the config file those add columns to

VoIP calls doesn't work in different networks (Using PJSIP and Kamailio server)

痴心易碎 提交于 2019-12-10 11:33:23
问题 I have setup kamailio 4.2 on an azure instance as server and for client I am using PJSIP library for Android and iOS applications. The voice calls seem to work well when both the devices are connected to the same network, however, either of the device connects to a different network (or when both the devices are in different networks), they are able to register on SIP server, and even call can be triggered and accepted between both the devices but there is no audio heard on either end. * I

VoIP calls doesn't work in different networks (Using PJSIP and Kamailio server)

旧城冷巷雨未停 提交于 2019-12-06 10:48:12
I have setup kamailio 4.2 on an azure instance as server and for client I am using PJSIP library for Android and iOS applications. The voice calls seem to work well when both the devices are connected to the same network, however, either of the device connects to a different network (or when both the devices are in different networks), they are able to register on SIP server, and even call can be triggered and accepted between both the devices but there is no audio heard on either end. * I have even setup rtpproxy. Can anyone please help me? Thanks in advance. Please check your IP address you

How to configure kamailio server with load balancing and asterisk? [closed]

Deadly 提交于 2019-12-04 08:44:49
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 4 years ago . I want to configure Kamailio server so that traffic will be forwarded to other four asterisk servers equally. It is working fine with a single asterisk box but I am unable to forward a call to another asterisk box. Here is the kamailio.cfg that I am using. #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #

How to configure kamailio server with load balancing and asterisk? [closed]

﹥>﹥吖頭↗ 提交于 2019-12-03 00:45:16
Closed. This question is off-topic. It is not currently accepting answers. Learn more . Want to improve this question? Update the question so it's on-topic for Stack Overflow. I want to configure Kamailio server so that traffic will be forwarded to other four asterisk servers equally. It is working fine with a single asterisk box but I am unable to forward a call to another asterisk box. Here is the kamailio.cfg that I am using. #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_ASTERISK # *** Value defines - IDs used later in config #

从Git安装Kamailio v4.3.x

限于喜欢 提交于 2019-12-02 18:38:10
###声明 本文内容主要翻译自 Kamailio v4.3.x官方安装与配置教程 。 ###安装环境和版本 操作系统:Ubuntu 12.04.5 64bit Kamailio:v4.3.x ###安装之前 先安装/更新所有依赖的工具/库 sudo apt-get update sudo apt-get install git gcc flex bison libmysqlclient-dev make libssl-dev libcurl4-openssl-dev libxml2-dev libpcre3-dev ###从Git下载源码 先为要下载的源码创建文件夹并切换到其下 mkdir -p /usr/local/src/kamailio-4.3 cd /usr/local/src/kamailio-4.3 下载源码 git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio cd kamailio git checkout -b 4.3 origin/4.3 注意 如果你的git版本不支持 --no-single-branch ,将其去掉即可。 ###调置Makefile 生成用于创建Kamailio的配置文件,加入需要编译的额外的module,并且为了删除方便