Stack Overflow
Stack Overflow

Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

208 questions
1
vote
1 answer

Unable to connect Linphone with FreePBX environment

Problem I recently installed a FreePBX server(includes Asterix). I want to try and make few test phonecalls before proceeding the development of an application. I use the Linphone linux application to test this. Unfortunately a problem arises: …
Jouke
  • 459
  • 1
  • 7
  • 20
1
vote
1 answer

How to change To header URI from Kamailio?

I am using Kamailio 4.4. I would like to forward the request to a different port number of my endpoint. I have changed the destination URI and the INVITE correctly reached the new port. But the To header in the INVITE request has the old port. so…
ARUNBALAN NV
  • 1,634
  • 4
  • 17
  • 39
1
vote
1 answer

Linphone change "replace + by 00" funcation

i all i am making a project using linphone in account there is option of "Replace + by 00". Example :- if number the number is saved in our phone +91-12345678 after using this option "replace + 00" what it does change this number to…
Abhishek Punia
  • 133
  • 11
1
vote
0 answers

How I stop the incoming call when I am talking SIP ANDROID

actually I have this code on Android : file IncomingCallReceiver public void onReceive(Context context, Intent intent) { SipAudioCall incomingCall = null; try { SipAudioCall.Listener listener = new SipAudioCall.Listener()…
Darlyn
  • 310
  • 1
  • 2
  • 16
1
vote
0 answers

sip.xml : How to make two sip servlet run on the same web application

I'm driving a bit crazy because i'm trying to make a different version of the restcomm app click to call and I can not find a way to make two different servlets running on the server(Tomcat 8 telestax version). Specifically I wuold like to use…
Stefano Angieri
  • 23
  • 5
1
vote
0 answers

How do I forward fisrt n calls to one phone number and then next n calls to another phone number in sip voip?

I have a vonage business account in which I receive hundreds of calls per minute. I want to outsource my process to different organizations so that they can handle those calls. In this I want to make a system in which make list of phone numbers of…
Harmeet Singh Virdi
  • 74
  • 2
  • 11
1
vote
1 answer

Cannot install Flexisip (Linphone SIP server) on Centos 6.8

I am building Voice Call feature for Android app by using Linphone. On server (CentOS 6.8), I install Flexisip by tutorial on Homepage of Linphone wiki (https://wiki.linphone.org/wiki/index.php/Flexisip:installation). However in step 1, I cannot…
Quy Nguyen Vu
  • 181
  • 2
  • 13
1
vote
0 answers

Twilio Origination Call Issue - Intermittent incoming issue

I'm trying to setup Twilio Elastic SIP Trunking on my Asterisk/ Freepbx instance and have struggle setting up a reliable origination (termination works perfectly fine). Background - I've done all settings as per twilio guides, tons of forum posts on…
tipsytopsy
  • 92
  • 8
1
vote
2 answers

Asterisk and Sipp UAS

I'm trying to get sipp communicate with Asterisk in order to perform performance tests: I've been through these steps: 1) In…
Striker
  • 43
  • 1
  • 6
1
vote
1 answer

Library Files missing in PJSIP project and how to link library files?

I am using PjSIP for SIP calling. I have integrated PjSIP project as per the instruction on https://code.google.com/p/siphon/wiki/Compilation & How To Build and Compile PJSIP for Xcode, Using sample code IPJSUA to test?. But when I opened the…
Aakil Ladhani
  • 984
  • 9
  • 32
1
vote
1 answer

Trying to setup Asterisk for voice chat between website users with sipjs. But unable to configure DTLS certificates. getting "hostname: Unknown host"

I'm trying to setup Asterisk Voice chat for users with the Help of Sipjs follows the instruction given on SIPJS docs http://sipjs.com/guides/server-configuration/asterisk. Users are created and also connected. They can call each other through…
K Ravi
  • 729
  • 8
  • 25
1
vote
1 answer

Failed to execute goal on project sip-servlets-core-api

I am trying to make an SIP servlet, after a lot of research on Session Initiation Protocal(SIP) servlets I am now using mobicent's sip servlet project with TOMCAT...! I am following this guide to get my goal, the issue is that when I am trying to…
Arsal Imam
  • 2,882
  • 2
  • 24
  • 35
1
vote
0 answers

SIP Servlet Container Scalability test

I'm gonna to have a scalability and performance test on SIP Servlet Container. what I want to do is to measuring the memory consumption and response time based on the deployed Servlets on the Container. Unfortunately I'm not so experienced in SIP…
Milix
  • 107
  • 10
1
vote
1 answer

Configure Kamailio to allow sip user from sending message to anyone but a specific user

We have a Kamailio SIP server up and running with authentication. Now we want that a SIP User say abc@localhost.com can only communicate to a specific SIP User say xyz@locahost.com and not to all other SIP users that are stored in a database table.…
Sharon Nathaniel
  • 1,467
  • 20
  • 28
1
vote
1 answer

Opensips byes rejected by client by 481 during redirection

Server is a Opensips server 1.10.0-tls (linux). It can handle conversations to/from local stations and recently it has been updated to allow stations from external systems. It does this by changing the username, ip and port in the $ru if the station…
TsukikoValkyr
  • 21
  • 2
1 2 3
…
13 14