Viber sip asterisktrabajos
need someone expert in asterisk to setup an outbound ivr survey for me
I am looking for someone to configure a small system (4 x Cisco 8811 + ISR4330) to be used as both an internal phone system and to be SIP-trunked to an external carrier. I would like to have it been set up to use the web-GUI afterwards if we need to do smaller modifications. Specific logo/text on the screen, speed dials etc. We will set up an online computer with teamviewer etc and connect everything to a temp network for your access.
We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party projects
We need a FreePBX expert. We configured a new asterisk appliance on Microsoft Azure Cloud We have issues with voice
Development Locally Debugging and testing via Remote Desktop SOC SIP NRF9160 Stage 1 Tasks 1 Read 8 Analog Inputs from MAX11610EEE 2 Read Digital Inputs BATTERY P0.29 / Debounce 500mS 3 Display Analog Input Values and Digital Input on OLED SSD1306 Display 4 Print Values over Serial Stage 2 Tasks 4 Send values via MQTT / over Ethernet ENC424J600 5 Send values via MQTT / over Cellular
...(whatever solution is best). The system works in two ways: 1 (attached graphic). Calls will come in through our DID. Each call is authenticated through our API (handled by a separate developer) and then each user is placed into separate channels after API authentication. The audio stream from the caller will be listened to and processed by our audio tools. The calling system is setup through Twilio SIP trunking. 2. When each user is in a webrtc channel, allow for the ability to have one or two users join via webrtc clients such as our mobile app, browser, etc. (another developer will handle client development). We are NOT building a conferencing app, but we need the ability to accept calls and then later perhaps have a client join via webrtc. This is all strictly on voice, ther...
this project is about building a script that will read from various discord servers messages from a list of channels under specific criteria and filters and then take some action. The action will be like send SMS, make a call or skype call or voip, send viber/whatsapp, skype Source code is required. This can be created to run online or as a script on a local PC You can use any programming language you like but python is mostly likely the most suitable.
We need expert on Flutter(more than 3 year ) developer expertise on API and XMPP
We have an existing hosted Asterisk server and use icabbi taxi dispatch software. I would like incoming caller ID to be screened to see if they have an active booking, if so give the caller an update on their booking. I would also like to give the customers an option to either call their driver, be passed to the call queue or cancel their booking.
Xmpp/Ejabberd/Flutter/Android/IOS we need integrate on chat flutter project
We need developers on skill webrtc voip mediasoup sip ejabberd
Hello!! My name is Shatira. I am the proud owner of Creative Arts by Sweetz LLC. My business is all about art. I create paintings and I make resin epoxy based products. I also am looking to host sip and paint parties as well. I am looking for someone who can make my vision a reality with my website. I want it to be welcoming and very professional. Thanks everyone so much and I look forward to hearing from you all.? below are some photos of some of my work but please feel free to reach out to me if more information is needed!! This will be my first website ever and I want to make sure im 100 percent cooperative to make that happen!!
i m currently using asterisk-FreePBX for inbound and blast outbound calls Asterisk is getting stuck by approx 400 inbound calls looking for someone to help me Setup a set of servers joined with kamailio or any other alternative to help me get to 2000 simultaneous calls on inbound and 1000 on outbound
Issabel 4 + SPAconnector 1.4 + VtigerCRM Installed. Just need to get Live POP up, Agent Creation and Asterisk CDR details, Recordings, Campaign in CRM. In Brief, Full Asterisk Connection with Vtiger CRM
i want to setup asterisk freepbx on cloud servers but i need it to handle approx 20000 callers and maybe expend it but each server on asterisk as i understand cant handle more then 400 calls so i m looking for a way to combine servers
Set up Asterisk PABX and integrate with proprietary CRM. Asterisk to have web interface to manage DDi routing and ring groups. API for CRM integration will be available. An option to switch easily between desk phones and softphones
webRTC janus-gateway SDP noSIP /sip plugin setup.. need to setup a call from a SDP Offer connection.
I need an Asterisk administrator who can configure two SIP trunks, modify the dial_plan, pjsip files, extensions files etc to an existing Asterisk build.
I need to connect SN4130 patton gateway to an Asterisk
Google Assistant webRTC developer This will be 2way call SIP / WebRT job
i have a openwrt installed router like tplink 3420 v5 now f...only 1IP rest of 7IPS free like 23.23.23.3- now my france router have connected my linux PC so i want to issue a static IP to my linux server. and my linux server have to be get this IP from server. now any one from the world get access on this IP: cause its an public IP . when my linux pc get public IP then i can run any service as i want like SIP=5060 web=80,443 mysql=3306 etc like that. i dont want any port FWD service. i want my local linux pc have to get full real IP from my server. so when my linux pc offline i can see ping down when online it will comes up ping up. if you can make it let me know. if you want to use any other vpn tunnel you can suggest me but not openvpn.
...(Change language (using flag icons). - Discussions will be made only by using voice communication (whatsapp, viber, messanger or skype) Skills Requirements: - 5 years of experience in building commercial B2B websites - 2 years of experience in JS - 5 years of experience in SEO - English speaking Please send us the following: a) A link to your best website (only one) b) If you also did the SEO for that website than add the keywords that are used to find this website (in a.) in google. c) A quotation that will include the following: - Price - The services included in that price - Milestones - Starting date - list of things that we should supply (logos etc) - whatsapp number or viber number or skype id Notice: - The full payment will be paid only at the end of...
Our company is developing its new home automation system and among the novelties it wants to introduce the direct management of SIP - VoIP video intercoms (through the local Ethernet network). The home automation system is written in Python3.8 and uses PyQt5 as a user interface (a knowledge of classes such as QMediaPlayer is therefore essential). The hardware is a PC-All-In-One with Intel i5 7350U 16GB RAM 128GB SSD processor, the operating system used is Linux distribution Ubuntu 20. We are looking for experienced programmers in this area to assist one of our technicians in the development of this part.
We need to configure Issable pbx (Like extensions, IVT, SIP trunk count). We have the diagram ready. Please quote a price if you can take on this project -Monika
We would like to make a 3D janitor model, based on our logo colours, so we could use for animation and viber stickers. Something like wolt mascot
A group of people delivering the natural flavours of coffee by Natural process, dry process, unwashed, or natural sundried processing that usually involves drying coffee cherries either patios or raised beds in the sun. This process, which can take 3-6 weeks normally, is the more traditional method of processing coffee. You can have that natural feeling every morning you wake, take a sip of Cafe Cabin and become fresh from the inside-out of your body.
Hello, I have the following error [2021-12-10 10:47:12] ERROR[2212] loader.c: Error loading module '': /usr/lib64/asterisk/modules/: cannot open shared object file: No such file or directory 5442 [2021-12-10 10:47:12] ERROR[2212] loader.c: Error loading module '', missing dependency: res_phoneprov 5443 [2021-12-10 10:47:12] ERROR[2212] loader.c: res_config_sqlite declined to load. 5444 [2021-12-10 10:47:12] ERROR[2212] loader.c: res_stun_monitor declined to load. 5445 [2021-12-10 10:47:12] ERROR[2212] loader.c: res_xmpp declined to load. 5446 [2021-12-10 10:47:12] ERROR[2212] loader.c: Declined modules which depend on res_xmpp: chan_motif 5447 [2021-12-10 10:47:12] ERROR[2212] loader.c: chan_ooh323 declined to load. 5448 [2021-12-10
I need to use linphone or blink or any open source softphone and to make some small GUI customization, adding BLF buttons and remote configuration (user will type specific code and will get all configuration from website) Then pack it all back to windows & mac installation. The one will take this task will continue to next project to do the same for mobile android and ios
We are looking for a freelancer to add inbound PSTN call support. Outbound PSTN call support is already implemented including call transfer and call conference. Please review the attached sheet for requirement. Please let me know the timeline and quote.
Hey Omkar, Derek here, I tried to figure this out a few weeks ago and forgot about it. I think this is the way I can make payment to you?
...established players is to follow a Lean Startup approach of building minimum viable products and then iterating based on customer feedback. We run two major platforms: 1) Life insurance () 2) Crowdfunding () Our Development Environment We use Angular, PHP and Java as our primary languages and open source tools that include October CMS, SuiteCRM and Asterisk. All our infrastructure is AWS based and we are committed to Automated Testing and Continuous Deployment. We follow an agile methodology combining aspects of XP, Scrum and Kanban. Our developers mostly work remotely and choose their own hours. You will be required to do daily updates on Trello/Slack but the only time you need to be available is 09h00 (GMT +2) for daily standups....
Hi Freelancers! I'm looking for a simple NodeJS application which takes an inbound call channel via the Asterisk Fast AGI/EAGI (Likely using the NodeJS package ding-dong, but any NodeJS AGI package will be fine) and then processes the channel via the Vosk API NodeJS package. The program should then: 1. Print that speech to text via the NodeJS console. 2. Within the current converted text utterance, look for the keyword - which will be 'chocolate' 3. If the keyword is found play a wav down the channel using the stream command then return to the dial plan (The wav needs to played via NodeJS, not using the dial plan - any wav file will do for testing). 4. If the keyword is not found, return to the dial plan 5. If no utterance can be detected for 5 seconds, return to th...
i have already have freepbx setup it needs to be configure to to work with twilio sip trunk send and receive phone calls Setup when call give you announcement options then customer can press in 1-2-3 go to different number I have number from twillio and elastic sip trunk from twillio
We are to start an IPRN promotion business. This means an initial setup (server, Asterisk, FreePBX, MySQL, dialer). Additionally, since we are mainly sales people, we are looking for a long term cooperation : manage the (automatic) calling activity, shoot troubles, control statistics. Please quote initial setup. We will discuss long term afterwards.
I need to build a software which supports my clients and connects them with my employees, I am looking for people having strong skills in Python, Node.js, My SQL or MS SQL building a CRM based web application with Data base Management The software Includes API Ping post, Multiple user access, Voip/SIP call, Subtitles display, Artificial Intelligence and Machine Learning. I need a Person/Freelancer who resides in Tamil Nadu India.
I would like to leverage existing FreePBX with GSM Gateway to manage inbound phone calls and outbound phone calls along with Exotel APIs for call concurrency. Freelancer must be able to setup dialplans, contact books for SIP extensions, call masking, Dynamic IVR, CRM and update the user details from School ERP through swagger API
Need an experienced Database integrator , to work on goautodial platform which is based on centos , asterisk , mysql , phpmyadmin, Well known about telephony system project will run within 2 weeks.
Hi, Given: SIP-Server A (IP A) SIP-Server C (IP C) OpenSIPS B (IP B) Calls can be from A->C and C>A. A->B->C C->B->A OpenSIPS 3.2 (B) in the middle should route the Calls with a RTPProxy. OpenSIPS should generate CDRs.
Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show al...pay clients for their incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" includin...
Hi Everyone, I'm looking for someone, who has experience in developing messaging, audio and video conferencing applications like Skype and Viber. The application would be using Agora API and many other microservices. I'm looking for people who have experience in Angular JS, Flutter, Android or IOS applications and even backend developers like SQL and NOSQL. The payment is based on weekly deliverables. My requirement is urgent. I trying to develop a prototype until 15th Jan 2022. Interested people kindly message me with your CV. Regards, Shreyasta Samal
I want to develop a mobile application that is going to function like a messaging app such as viber/whatsapp/messenger. I have the design layout and a to do list of things. I need a developer to code it and provide me with the final results.
update python function, by adding 2 more parameter, Need to know Mysql asterisk is option
Looking for 3 different t-shirt designs representative of our company. Need 3 very cool designs to put on our merchandise. Front and or back design. Sip-Line, is a hands free beverage carrier that keeps your beverage upright and does not spill. Our product is used in the fishing, hunting, camping, tailgating, hiking, and boating industries. We would like designs that include our product some way. Please reference our website and Instagram for ideas. , @sip_line
INSTALLATION OF VICIDIAL ON A SERVER | So after you install the asterisk on your laptop you can use this server and forward to the other server help to install voip server with basic installation. help to install voip server with advanced installation. Asterisk Freepbx Elastix Vicidial
Hi We have a Claris Filemaker application. Currently using a paid-for SOFTPHONE, which we want to transition to a cheaper version (we are a charity) I am hoping to use LINPHONE softphone. Just need some help with how to get the application to make calls using Linphone - it has installed, and the App calls up the default SIP - but calls are not initiated. Can you help?
i Have Avaya Server edation R11.1.2( VM) with Remote Worker and Sip trunk connected to ASBCE R 8.1.2(ASBCE CORE PORTWELL CAD-0230) it was working fine but now when im traying to call from remote workerto outside or local ext its givin me call failed the user not found , some times if i close the Avaya IX work place and open it again its working , in ASBCE i got two error 401 unuthorized , and 404 not found .
I need to use linphone or blink or any open source softphone and to make some small GUI customization, adding BLF buttons and remote configuration (user will type specific code and will get all configuration from website) Then pack it all back to windows & mac installation. The one will take this task will continue to next project to do the same for mobile android and ios
We are launching a new newsletter by the name of "Evening Sip." We are looking for some creative 500x500 (though we won't disallow other dimensions, but they'd better have a good reason for the deviation) logo that is colorful, fun, and definitely plays on the "sip" portion. This will be an evening newsletter that covers a little bit of everything across multiple news categories. Please provide us with your best logo ideas, and feel free to bring your most creative ideas to the table. If you have any additional questions, don't hesitate to send them over to us for clarification.
