Plivo sip freeswitchtrabajos

Hi, I have finance blog in Hindi and want to have following calculators for my website: a. Step Up SIP Calculators b. Target Amount Calculator c. Goal Based SIP Calculators d. Provident Fund Calculator e. Retirement Planning Calculator f. Asset Allocation Calculator Thanks, Rahul

We have a web based IP phone that utilises version 0.14... we need the javascript code changed to utilise the latest support version of , which is 0.16, there is a fundamental change in the API interface and without spending more time than available, we cannot make the changes and get them to work. We would provide copies of the current source files and a test SIP username/password so you can fully test and demonstrate to us it works with the new code before we update our current production code. If you do not have experience of version 0.16 and above, please DO NOT bid on this project. Thank you.

We are a 4 year old telecom company and we’re looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS React These skills are optional: Facebook APIs We will offer long term work every week but only if you prove your skills in the first week or so. We are only looking for serious developers and to prove that please fill the attached document, upload it anywhere and add the link to the project proposal. Your proje...

Ich habe einen MacOS Catalina und benötige Hylafax um Serienfaxe zu versenden. Auf dem Mac soll das Paket T38Modem kompiliert werden unter root per SSH Zugang für Sie remote, bzw. sollten Sie per VNP oder Teamviewer zugreifen und darauf arbeiten, bzw. sich auskennen mit Hylafax und T38Modem. Ich versende die Faxe mit einem VoIP Account bei SIPGATE und SIP über einen SIP Server, sowie mit der command-line per sendfax 012344535234 beispielsweise dann, wie unter Linux.

We need to build a WebRTC platform that can bridge H323/SIP videoconferencing systems. Must work either on premise or as SAAS and allow integration to other SW platfforms. Development in Angular is prefered.

O projeto consiste em um balanceador de tráfego SIP. Os destinos devem ser recuperados de uma tabela de banco de dados MySQL. Uma ação (executar uma URL ou update no banco) deverá ocorrer se algum dos destinos retornar um código específico (Ex: 602). Não será necessário verificações de segurança (acl, etc) pois o ambiente de produção será rede local. Não será necessário adicionar serviços de RTP pois os áudios serão fechados direto para o IP de destino, resultado do balanceamento. Por ser parte integrante de um projeto, não será necessário nenhum tipo de interface do usuário. Preferência por instala&cce...

sip to whatsap, viber, telegram, signal getway

Applicable for Android App (reactnative, mongodb, nodejs) and website: 1. Hiding network details and just showing and describing network strength Excellent, Very Good, Good, Poor etc. But it should not affect other options anywhere. 2. Customising color Jitsi Option Menu as per our chosen design/color 3. Meeting Creation Link ...Server hosting independent and customised instance of Jitsi 4. Random name creator 5. Replacing Jitsi name with ouor brand everywhere in platform. For example if using our meeting from Muscat, someone got Jitsi error. So all error codes to be replaced with our error codes/numbers and explanations. Depending on if above is done in agreed cost, would consider for autoscaling, load balancing, Jigasi+SIP+VOIP, Recording upload to a link etc. I am in Kolka...

Hello, I am looking for an expert who can help me with sending VOIP pushnotifications for iOS (iPhone) devices using Asterisk / FreePBX environment. I am using Linphone SIP mobile client and I need you to help me configure VOIP Push service for the incoming calls. If you have experience in what I am talking here, then please bid on the project. Bid on this project only if you have experience.

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We need a softphone that will work with our pbx systems. It needs to work on 3 platforms; IOS, Android, Windows. You can see the detailed information below; • “Sip Server – Username / Password” based login screen - (We require this for the current project. After this project we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must able to ...

We want to use Asterisk PBX with Avaya AAEP. The idea is to transfer the call when it reaches the IVR to another PBX (Asterisk) installed locally. Requirements - * Integrate Asterisk PBX with AAEP for SIP transfer * Call transfer can be both bridge or blind. Both, from Avaya to Asterisk and back should be supported * During transfer metadata like caller_number, call_language, etc should be passed

Looking for a PHP developer with the following skills to join team. ** NO AGENCIES PLEASE ** - Full understanding of MVC Frameworks like: CakePHP, Laveral, CI - Understand of REST API - Twilio SDK - Telesystems like SIP, SMS - Ability to work hours (600 - 1400) US Eastern Time. - Team Player. Able to take direction from Project Lead. - Able to communicate in SPOKEN English. - Good Customer Service skills. First 2 weeks will work as trial @ $10/hr. Negotiable there after.

Need someone with specific experience configuring Ricoh Copiers for IP Fax using SIP.

I run painting parties on the weekend. Virtual Sip and Paint ;) But I have two full-time jobs during the week - and a family. I love painting and instructing paintings - but I need to promote the events two weeks in advance - so I need to have a painting composition two weeks early - but I am already so pressed for time. So I need a creative hand. A ghost-painter... y'know like a ghost-writer but for paintings. These can be made digitally or traditionally. ...THEY MUST BE YOUR OWN UNIQUE IDEAS (not from Pinterest. smh)!... BUT keep in mind, the classes take place over 2.5 hours and use 16x20 canvas and acrylic paint, so the composition must be feasibly accomplished by a lay-person (non-artist) with acrylics in that amount of time. It should be something you could easily...

I need an Android application creating which can act as a GSM gateway and SMS gateway. It needs to: Voice gateway: 1. Connect via SIP (or WebRTC) to my voice switch, which is hosted on the internet and not local to the phone 2. Accept calls from the voice switch server and place call out over the GSM connection. 3. Provide two way audio for the caller/callee Call generator: 1. Accept commands to place call on GSM side and make call 2. No need to connect audio to voice server 3. Play recording (or make DTMF noises) from phone over GSM call 4. Hang up call when required SMS gateway: 1. Connect via smpp to my SMS server (or use HTTP polling to a HTTP endpoint) 2. Accept SMS from my SMS server via SMPP or HTTP polling and send it out over the GSM network 3. Accept SMS from the GSM s...

...DRINK CUP AND FOOD HOLDER IN ONE AND THE MAIN PRODUCT IS FLAVORED FRENCH FRIES. AS A TWIST, WE PAIR IT WITH A DRINK LIKE SOFTDRINKS OR A FRUIT JUICE. THE IDEA IS CONVENIENCE AND MOBILITY WHILE EATING YOUR FAVORITE SNACK BECAUSE OF THE IDEA OF HAVING A DRINK AND A FRENCH FRIES IN JUST ONE PACKAGING. WE WANT YOU TO CONSIDER TRYING TO DESIGN THE BRAND NAME WHICH IS SNACK & SIP WHICH EMPHASIZES THE LETTER “I” IN THE WORD SIP AS THE STRAW. BUT IF YOU HAVE MORE AND BETTER NAME THAT SUITS WITH THE CONCEPT THAT WOULD BE GREAT! WHAT WE NEED IS SOMEONE THAT CAN CREATE A FUN, YET VERY RELEVANT LOGO FOR OUR BRAND OR NEW BRAND NAME OF YOUR IDEA. FOR THE LOGO CONCEPT, WHAT WE WANT IS FOR IT TO BE PROFESSIONALLY LOOKING BUT NOT BORING. WE WANT IT TO BE CARTOONISH, WITH...

I need to customize a Java open source sip auto dialer. The person needs to be an expert with a clear knowledge about voip. the opensource with the source is at GitHub: Basically, I want the software to perform a call hold, dial on the same port another call, then upon the success of that call, bridge that call with the call in hold. The software will dial to an IVR and the purpose of this job is to be able to use the same port 6 of a maximum of 6 calls, that is allowed by the sim card into the gsm gateway. so, we need to add that feature to open source. If you feel you can do this job, please send me feedback to discuss price and timing, we need this asap, thanks!

we need an expert in linux pulse audio or jack tools to configure a virtual audio between audio incoming from usb to a sip client in the linux

I'm in a need to setup a SIP trunk that I have into my Voice Cisco Router 2901. The Voice Router is behind a Cisco ASA device. I do have access to the ASA device to setup NAT or any other configuration needed. Thanks

Hello I install a FreePBX with Custom Destination. It is connecting to my Provider using 7 IPs for SIP any there many IP for RTPs. The SIP Trunks are UPnbut when I call a DID, the call reach the FreePBX but there is no voice. I need someone help to trroubleshoot that. Thanks

To set up installed Asterisk PBX stable 16 qnap x86 SSH small set up 4 sip 5 phones 5 softphones GUI - web admin simple voice missed number to email debatable option

We are looking for a custom-built SIP Softphone (PJSip Asterisk extensions) project using .Net 5 and C# to with the following features: 1) Source Code in Visual Studio project using .Net 5 for Windows Application 2) Downloadable internal directory/contacts 3) Simple UI 4) Minimizable to Tray 5) DTMF and Multi-line with hold, conference, transfer 6) Configurable ring notifications 7) Cannot use proprietary/paid license components 8) Completely configurable from web endpoint (can be XML/YAML/text for the extension, server, etc) 9) Code standardly documented in English

Looking for art instructor with personality, group training experience, experience with acrylic paint - water based paint and canvas. Must be able to work with beginners in a creative environment with a smile. Opportunity to present art portfolio in the artsy village of Nyack NY. Budget is open.

Hello, Few call center employees need to work from home, so VPN need to be set up to connect them from home to head office network to make & receive phone calls. Input data: - We use PRI line with 100 numbers, out of each around 5 need to work remotely. - PBX is Asterisk-based; also Yealink, windows server 2012, Call center agents use X-lite old version as free SIP-phone tool. If needed, there is DyDNS address of the head office, as head office IP address is dynamic. Server & PABX map is attached. Time to finish project : 2 days + 1 day testing. There is no VPN existing in head office currently. Call center laptops at home for connection with VPN have Windows.

I need an iPhone app. I already have a de...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

Hello, the project is based in develope a configuration into a CISCO ASR1002 to customize a SIP/SS7 signiling switch. The SS7 signiling is received in TDM/E1 using a Channelized STM1 Interface connected directly to the Operator, inside the STM1 the first 8 E1's has the SS7 link & CIC's. The system will have to convert to an ITP / Point Code for SS7 signaling as a part of the final solution to be used as a SIP SBC or Sigtran point. Components: 1x CISCO ASR1002 1x Channelized STM-1 port (Cisco 1-Port Channelized OC-3/STM-1 Shared Port Adapter), the stm1 has 8 E1's. 2x SS7 Link (OPC/DCP/ADJ Point)

...certainly be a plus, but is not a requirement. The type of jobs that we'd be expecting the freelancer to perform are: 1. Listen through the un-edited version of each new episode and annotate time-stamps for editing (e.g. delete section from 1:39 to 1:52). The positions for cuts will be obvious, as it is usually sections where the podcast host and guest discuss logistics, take a break to have a sip of coffee etc). Each episode will be roughly 60 min long. 2. Annotate the timestamps where images would be added (we'll provide the images, so it would just be necessary to find the right timestamp when that image should be shown in the video) 3. Annotate timestamps for sections that are "highlights". Those will be cut together separately for a shortened summary/h...

Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.

The app is 2 pages (+ login page) Page o...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

I would like to get the SIP trunk configuration for goTo and Nextiva sip provider

Hi I'm using fusionpbx 4.4 vs freeswitch 1.8.1. I having a problem with 30-50 cc but cpu haved load over 70%. Server 12core 24GB Ram.

A list of apartments and their owners on a screen. By pressing a phone button, you will call that person. If the person wants to open the door, he presses *9 or something like that. All on a web page. There are JavaScript sip clients. The phone part is by asterisk installed on the same raspberry pie using google voice.

...page, whereby the sub-brands will sit under the motherbrand. The objectives are : 1/ to promote the brand awareness 2/ to increase market share & promote sales 3/ to have more engagement with people The new branding will focus on: - Having one singular brand to promote all the products - Tone of voice will be more human then informative, 1st person conversations - Setting a new positioning: A sip of Life [in french: Gouter a la Vie] What we imperatively need in the offer: - creative ideas for the year 2021, ex. an idea for Water Day / Environment Day etc. - improvements from the actual Facebook page [Vital Eau de Source] - a roadmap calendar, which can be referenced by all stakeholders - a good management tool which will suit our agency, the client as well as the community...

Need to Upgrade Freepbx from 13 to current and install: SIP Poe Mullion Video Keypad Intercom 8039 8063 SIP Interface Module (IP Relay controller) for Algo door control only. 8063

Необходим человек который оперативно сможет рассказать как и что настраивать в VOS3000. В первую очередь необходимо установить SIP Header Diversion для поставщика услуг

we need an android expert to develop an app that when activated it will 1 to . rout the audio (speaker and Mic ) of a sip client to jack 3.5 headset 2. rout the audio (speaker and Mic ) of the cellular gsm calling to USB headset 3 . this audio routing need simultaneous ( in the same time you can hear and talk both conversation ) the budget is 250 dollars

white paper about the social issue investigated for SIP. The purpose of the SIP white paper is the following: (1) capture the main essence and challenges inherent in a prevailing social issue and (2) propose an actionable recommendation to address the challenges contained in (1).

Setup Plivo SMS so we can send and receive text messages to prospects, via Kartra.

I am looking for a developer to write a code for IOS/Android push notification for VitalPBX SIP services to work with Zoiper or Linphone Project Cost US$ 100.00

Real Estate Company seeks positive, accountable, skilled, motivated, self-driven individual with experience in sales. Great voice on the SIP phone crystal clear microphone/equipment and high speed reliable connection for making calls and must love speaking to people on the phone, silent working environment for professional calls. Sales experience, preferably real estate related, but will train as necessary for knowledge and understanding of contracts preferred but not required. " Must be skilled and sound good on the phone with some understanding of the US residential real estate market " Our payout two types, you can choose anyone. A. Hourly Based Payout. B. Commission Based Payout. ( A. ) Payment terms per hour : As long as you stay connected to a call, but not for how...

I have Dell R630 5TR Raid-5, 64G RAM housing Ubuntu-16.4. it is integrated with my SIP Soft switch. We use the Datto backup service that allows us to install a Datto Agent on the Ubuntu servers and the Datto agent software performs hourly backups. Three other U16.4 servers back up just fine. According to Datto support, they have seen this before where during the FTP transfer from the Ubuntu machine to the Datto server, data becomes fragmented. They claim the fsck needs to be run on the partition. I have attempted to do just that, however, partitions need to be dismounted and few other preemptive steps need to be taken. We are a small business, Mitel phone system distributor hosting our own VOIP platform I don't have any Linux-trained techs available so I would rather pay a kno...

HI, I have a IP phone yealink - T21P-E2. I need to know its SIP credentials, mainly the PW. I have local access to the device and can give teamviewer or anydesk to retrive the info.

we need an expert in Android to create a small script that will rout audio cell phone call to USB audio and in the same time to rout sip Client (softphone) audio to Head jack the the cell phone and the sip need to work in the same time our budget is 250 dollars

I would like a multi-class SIP tutorial, at first with the most basic concepts and increasing the level with each class, begining with a basic level and finishing with more advanced concepts, to learn and understand completely how working SIP networks. The minim total duration of the videocourse with all of classes is 30 minutes.

I have to connect an 3CX pbx to an T-System SIP-pure Trunk (sip-trunk based with 3-way authentification). I want to use kamailio as sbc - but i have no experience with kamailio sbc-config. I need someone to write an to archive the following: - register to T-System Trunk (3-way auth) - connect to 3CX (sip-trunk) - rtp engine to convert g711 - t.38 and vice versa - 3cx should have the option to use re-invite and replaces - T-System doesn't support them directly wfr Michael

Simulate SIP vulnerabilities that include DDoS, DoS, password guessing, Man in the Middle attack, and SQL injection using Omnet++ in VoIP architecture. Take a video of each step of the process. The video should have no voice but should be clear in terms of pictures. Also, include how to mitigate each attack mentioned above in the simulation and capture it in the video.

i need the numbers in twilio to be sip accounts and to create an ivr for incoming calls.

Hello, I have installed an Openvox UC 501 Asterisk Based IPPBX at my home. I have a BSNL FTTH Connection with a Landline Number. The landline is currently working through a POTS port on my Syrotech ONT Modem. I want the Landline (VoIP) to work as an SIP trunk on my IPPBX. I live in Raipur, Chhattisgarh, India. Thanks.

Dear FreeLancer, i have fresh installed 3CX, i need to configure the Sip trunk IP Authentications based, outbound/inbound calls.

Hi, do you have good experience with Kolmisoft M2 softswitch based on freeswitch ? we are facing wrong disconnect cause. Regards,