sip-server

retain original caller id on Call transfer on asterisk

由 廉价感情. 提交于 2020-01-23 16:52:12
问题 I am running a B2C outbound Campaign on VicidialNow C.E 1.1 as Asterisk Server / SIP Server . The call is made from server to customer and connected to agents waiting for calls. The agents transfers the call to third party (not a blind transfer). The 3rd party sees the Caller ID of agent. Now, what I want is to display the caller id or the phone number of the customer to the 3rd party. I Googled and searched over SO, found this sendrpid=pai to add on sip.conf file. but this functionality only

Error while building Linphone for windows

由 情到浓时终转凉″ 提交于 2020-01-01 04:31:06
问题 I am trying to build linphone for windows by following the instructions in Readme.mingw using MinGw/Msys. There is no issues till the download of belle-sip package. When I run ./autogen.sh I am getting the below error. Generating buildipts in belle-sip... + libtoolize --copy --force libtoolize: $pkgltdldir is not a directory: `/mingw/share/libtool' + aclocal -I /share/aclocal Can't locate Automake/Config.pm in @INC (@INC contains: /mingw/share/automake-1. 11 /usr/lib/perl5/5.8/msys /usr/lib

Linphone core listener not receiving incoming calls

由 前提是你 提交于 2019-12-25 02:29:56
问题 I was trying to add sip incoming calls with linphone sdk, The registration is successful and I can make out going calls and the call status is logging as expected, but I am not able to receive incoming calls. I am using intent service to handle connection. Here is my code: protected void onHandleIntent(Intent intent) { String sipAddress = intent.getStringExtra("address"); String password = intent.getStringExtra("password"); final LinphoneCoreFactory lcFactory = LinphoneCoreFactory.instance();

How to generate big number of SIP requests

由 淺唱寂寞╮ 提交于 2019-12-24 09:44:26
问题 I need to test an application that processes SIP requests. For now, I want to test the performance of the application, so I need a way to generate a big number of SIP requests. I know there are tools for this (like SipP), but I don't know what is the maximum number of requests that a single computer can really send in a particular time interval. I never done this type of test, i need help. Thanks 回答1: Well sipp can generate requests pretty quickly and if you're testing call set up and tear

Why is dynamic real time not recommended as per asterisk?

由 我是研究僧i 提交于 2019-12-24 00:33:30
问题 In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like func_odbc." 1) Why asterisk is not recommending dynamic realtime extensions? 2) How to do static dialplan with data abstraction using tool liek func_odbc? My requirement is having have more extensions (in this case mobile number) coming up, how can I

What's the encoding used in SIP or in SIP ALG in Firewalls

由 本小妞迷上赌 提交于 2019-12-13 03:45:46
问题 I have written a Python 3 script to emulate SIP call transfer event. The script was working successfully when tested with a SIP server sitting on the same network. When I tested with a SIP server sitting across a firewall, all data sent by the script to port 5060 of the SIP server is blocked by the Firewall. Data sent on any other port than 5060 are allowed across the Firewall. This made me presume that the packets are blocked by SIP ALG protocol running on port 5060 of the Firewall

How to get call info from Kamailio

由 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

Genesys Platform : Get Call Details From Sip Server

由 一笑奈何 提交于 2019-12-08 02:53:34
问题 I want to get Call Details from Genesys Platform SIP Server. And Genesys Platform has Platform SDK for .NET . Anybod has a SIMPLE sample code which shows how to get call details using Platform SDK for .NET [ C# ] from SIP Server? Extra Notes: Call Details : especially i wanted to get AgentId for a given call and From Sip Server : I am not sure if Sip Server is the best candiate to take call details. So open to other suggestions/ alternatives 回答1: You can build a class that monitor DN actions.

SIP over websockets to true SIP

由 ￣綄美尐妖づ 提交于 2019-12-07 06:20:10
问题 I'm trying to implement a sip server for connecting to from an HTML sip client(made using sipml5). During my research into doing this I've come across sip over web-sockets which might be useful to me, however, I am unsure if a user agent connecting through sip over web-sockets to a compatible server would then be able to successfully make a call to some one using an incompatible server(i.e. calling from SIP over web-sockets to true SIP). I know webrtc2sip can be used for connecting to legacy

Genesys Platform : Get Call Details From Sip Server

由 十年热恋 提交于 2019-12-06 05:08:38
I want to get Call Details from Genesys Platform SIP Server. And Genesys Platform has Platform SDK for .NET . Anybod has a SIMPLE sample code which shows how to get call details using Platform SDK for .NET [ C# ] from SIP Server? Extra Notes: Call Details : especially i wanted to get AgentId for a given call and From Sip Server : I am not sure if Sip Server is the best candiate to take call details. So open to other suggestions/ alternatives You can build a class that monitor DN actions. Also you watch specific DN or all DN depending what you had to done. If its all about the call, this is the
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