Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.Hire Asterisk PBX Developers
*Please read to the end before responding* I'm looking to extend the capabilities of the GL-MT300A mini router (OpenWRT based) I want to leave its current capabilities intact, but enable SIP recording I have a similar setup on a Raspberry Pi using Oreka and inotify to upload completed audio files to my server Here is what I am looking for as a deliverable 1. Can actual wav files be produced on the GL-MT300A, maybe by using Oreka? 2. Can the stock firmware be extended to have a new "registration" page? All I really want to capture there is an email address and a registration token? 3. Can a file be uploaded as a base64 encoded object via an https: API? If 1. cannot be achieved, then I, know that pcapsipdump does work, so I can accept the pcap file as an upload if necessary That is the investigation piece If I can get a "yes" to the above, I'll be looking for a quote to rebuild the firmware with a) recording capability (eg Oreka or pcapsipdump), b) automatic upload to my server of the resulting files, c) the "registration" page, and d) automounting of USB drives that will be used for temporary storage of the audio/pcap files (the device only has 64Mb of RAM, which in terms of free space only gives 30 mins max of recording). This device can accept an SD card internally, so that is an option if it can be autoformatted and mounted The latest firmware has a nice feature that uses the side button to switch from DCHP mode to repeater mode, which is pretty key for this implementation Questions to answer in your proposal - If you don't answer them, you won't be considered What OpenWRT experience do you have? Have you developed on the GL-MT300A before?
I need urgently an expert Perl developer having good experience in writing Perl scripts in Telecom or Banking sector. The requirement of the project is that developer has to share some part of the code with us. We will go through the code and judge the quality of development. Once, we are satisfied, we will take the process to next level.
Necesitamos un Softphone hecho en Webrtc integrado con Asterisk, el Softphone debera tener todas las funcionabilidades tales como, Hold, Transfer, DND, Conferencia, Park, DTMF type select, etc. Tambien debe de terner forma de telefono, es decir botonera. Se debe proporcionar todo el codigo fuente documentado.
FOREMAN and KATELLO can be configured, to automatically bare provision a LAMP server running Wordpress into DEV stage, then promote the same server to Prod stage. You are free to chose the operating system and appropriate configuration management tools(Puppet , Chef or ansible which is recommended )
Project Description I need an application that runs on Android phones and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. If android Layers have a problem for this job, you can use replicans OS. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice vers General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the application from source Features - Route call from SIP to GSM - Route call from GSM to SIP (preconfigured SIP client) - Convert audio from/to SIP and GSM networks Application Configurations - Enable or disable the app - Enable to receive calls only on wifi or all networks - Enable to route calls from GSM to SIP (SIP address is configured through an API) - The number to send the call should be configured through an API that is supplied with the APP - Enable to route calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made
We are dealing with both outbound and inbound calls of Technical support of US. If anyone is interested [Removed by [url removed, login to view] Admin for offsiting - please see Section 13 of our Terms and Conditions]
We need to install Asterisk. The person required to give us consultancy on how to install and know how of the Asterisk. The person should be able to give us details of different tables in mysql Database.
The project to install Cisco Phone models as 88XX series & 78XX series to work with Elastix Freelancer has to do TFTP, XML file, a must t activate softkey buttons such voicemail, conf, Call History, Phone book, etc .. Test will be on 2 phone & to gurantee all phone we add to work in the same way of file or upload Once all working, Job can be considered as completed & Payment will be released
I am looking for someone that connects my Twilio numbers with my Xlite or Bria softphone. As far as I understand, a SIP registrar is being needed and OpenSIPs seems to be an appropriate solution. I am open for alternatives. A server is here.
Hello, I am running a problem with my existing project is not functioning properly. Need an urgent fix on this. Skills required: PHP , Mysql , Encryption and encoding algorithm Additional Skills: Asterisk , a2billing Please mention : " I am interested to do the job" in the subject line of the job application.
please read carefully before accept my job i have termination gateways work with CDMA network not gsm have issue that is when make call it run billing directly with calling before ringing or real connected this happen even i use my mobile for same network so is any one can fix by soft switch (Signal billing inside sip)
I have several 7861G handsets, a Mikrotik router, and FreePBX with Endpoint Configuration Manager. I'm looking for someone who has done this before to set up the configuration to make this work for us. Thanks
Sistema de mensajes masivos SMS así como llamadas por medio del sistema Elastix
Hi I need a voice over done please for some IVR - Person needs to be in GHANA Native Ghana Accent English language preferable, if you can do Hausa too then it would be advantageous then we can negotiate on this price But if you can do both then please quote me English Only Hausa Only Both Scripts are attached as below 1) Need each file split up 2) Someone with a Jolly Festive type Style! Thanks