OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO
Asterisk - CTI (Click To Call) API This will work on a Asterisk 13, FreePBX 13, and Asternic Pro 2.1.x platform. We need to provide a REST API for several actions e.g. a) Click To Call - e.g. /[url removed, login to view] authid = preset hash value to authenticate the user against. Return in JSON - Asterisk
...syncing for both server A and B. If server A goes down server B must take care of all the things. Like MX records DNS entry. 3. We have a newly installed Asterisk server. Need to configure the Asterisk server. Here are the following things you need to do. i). Configure all extensions. There are total 7 extensions need to create. Those are given below
We are KareXpert Technologies a HealthCloud...KareXpert Technologies a HealthCloud Platform where a doctor can make video call with their patients. We have our sip client(android), webrtc(webapp) and kamailio as sip server. We need an expertise for debugging and to give solution to make sip call work. Problem is webrtcToSip call freezing at web side.
...error when connecting to Video Call, we will be using the free sip service of: [url removed, login to view] The goal is to make the sample work with audio and video when making a call to a SIP client installed in a Mobile phone. the clients will be using also a free SIP account on that server. Audio calls go through the system fine
I need few answers for topic.I am basically networ...from Basic stuff, since i am new for this technology.I have interview ,so need to prepare for those answer. 1- Manager baghto bolla 2- Lte cdma 3-Handover reselections 4-Sip call flow 5-Earfcn downlink 6- Lte master information block 7- Types of registration in cdma 8-Tracking area update
I need some graphic design. To put on t-shirts. They are supposed to be envrionement themed.
...modeC=with capcha, customer loged in to see created sip account and code to copy and paste to their web page 6- modeD=if mobile telephone number is all ready in database say sorry and if email is all ready in database say sorry 7- mysql database name wsip 8- to create sip account will use asterisk database 9- delete account user we must delete from
I need an experience free lancer that can fix connectivity issues and Vicidial asterisk based system. If you don;t know Vicidial please don't even send your proposal. Long term project with some support needs on demand.
I need a SIP Client programmed in python. This software should be/have: -Both, the SRTP protocol and the ZRTP protocols are needed. -It should have video calling and graphical interface (GTK ...). -I also need to have the ability to configure 2 extra buttons (relays) for different purposes. -I also want to be able to add a logo inside the graphic
Prince Themed Stencil with Prince and his symbol
We need someone who can write an application to be used via the voice control of Alexa (Amazon) The voic...an application to be used via the voice control of Alexa (Amazon) The voice control needs to listen to s certain request and execute it then and this can then send in a sip instruction to a telephone switch to be answered by an operator.
- Routes calls to Route1 or Route2 based on 2 simple database que...2 - If Query 2 is negative we route the call to Route1, if positive we route to Route2 - Telephone numbers are 12 digits long, no special characters - Should work with Asterisk 13.14 and MySQL 5.5 - Should handle 5-10 calls per second without issues and need for special hardware
I have my own DIDs bought them from a site has only one option to forward the incoming calls and SMS to SIP URI. I need you to setup Asterisk and configure it to receive the incoming SMS and to get these SMS's through an API via my website so i can check them via my site.
I have the solution and all the call flow and everything is written i atacht part...everything is written i atacht part of the file the rest of the file is similar Just need to write in the correct syntax without errors in DIALPLAN so if you are good at asterisk dilplan it is very simple for you and if it is not simple for you . dont bid for it