Asterisk definity sip trunklavori

Hello Freelancers, We have linphone based android and iPhone application. Both apps are working fine. But the issue is background calling. Means If app closed then its not working. so we are looking for the developer who can help us to achieve it via push notification. We are using Asterisk based system. Thanks

This project is to setup a web SMS management software and gateway (Gammu + PlaySMS) along with Asterisk (freePBX) system that also uses the USB modems for calls in and out. The project must support SMS in and out with multiple usb modem dongles and sim cards simultaneously with calls in and out

I have attached a PDF with all the information to get started. I am looking for a VoIP Server that can be set up on my AWS Server and instead of normal phones connecting I am going to have some custom hardware made to connect to it for Two Way Radios.

I have attached a PDF with all the information to get started. I am looking for a VoIP Server that can be set up on my AWS Server and instead of normal phones connecting I am going to have some custom hardware made to connect to it for Two Way Radios.

I need someone that can remote in and reset the loop trunk cards.

We have a custom build of MicroSIP. This is used for Win/Mac/Linux (using Wine) devices. Source code for our custom build of MicroSIP can be provided. We would like a similar application for Android/iPhone devices. Audio codecs supported should be G.711 and G.729a. Video codecs should be H.264. We will maintain ownership of the source code for both platforms.

...phone numbers - users should be able to choose phone number through dial prefix) 2. We have SIP Trunks that needs to connect to the new Issable image. 3. Follow the image for the IVR flow and create the IVR flow 4. Creating Ring-groups 5. Configuring Standard firewall. Sip traffic should only traverse the firewall if the other side of the call is outside of the office or at a remote location accessible by VPN 6. Make new recordings according to the script provided. 7. Create a xml file for contacts which will be integrated to the system so that when contact added / removed from the file it will be automatically refreshed. 8. A shared voicemail for sales and tech accessible for all users 9. Check SIP connectivity to see if all the trunks are up after the whole setu...

looking for someone who can set up server with Indian SIP domestic calling service

looking for some one who can set up server with Indian SIP domesting calling service

Looking for a skilled developer to help develop custom features and apps to work in hand with FreePBX. This would be an ongoing relationship, as our customer base is constantly putting in feature requests. First project would be to develop an application that links to PBX users and utilizes SIP Trunks for sending and receiving SMS messages. Within the sending and receiving, we will need the ability to create groups for mass outgoing, but in a way that each end user is messaged individually so the responses only come to the sender and not to each member in the group. Additional projects will be discussed with the awarded developer.

to fix dial plan to do the following : 1) dial the trunk and ignore the first congestion from trunk because when we get congestion the call keeps flowing to the trunk so we want to ignore the congestion I get and keep the call when I put 2 dial command it works but I want to ignore that can we ?

Hello everyone, There is a FLUTTER project for which designs are already prepared. Please find the zip file attached with the designs. Need to only integrate this VoIP SDK: Here are some features we want from that SDK: -- Voice Call; -- Call Transfer; -- Conference Room; -- Record Calls; -- Call on Hold; -- Call History; -- Contact List; -- Settings: User information; SIP settings; -- Authentication with license Final output will be the Android and iOS project submitted to the store with the SDK integrated for all these screens. Source code will also be shared for analysis with the chosen candidates. TERMS: 100% Payment will be funded to Escrow, but will only be released once project is complete and approved by the

I need to integrate CyberArk and Asterisk -Issabel for password vault.

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DynDNS address of the head office, as head office IP address is dynamic.

I'm starting to use Accredible to generate the certificates for VoIP School. I want to create custom certificates and badges for the school. The current certificate and badge (I think they are really ugly) i did it myself. The badges will have the title Verified (Asterisk/FreeSwitch/SIP) professional I'm uploading the current certificate and one badge I did (as I said terrible) and the VoIP School Logo

we are looking for freelancer who is able to install and configure Asterisk with FreePBX and USB dongles for voip termination (mobile). If you have the required know how please do not hesitate to contact me and say which hardware requirement should we have and the duration do you need to finish the project and the price to do this.

Hi, I want to have an asterisk installed on the digital ocean with a customized caller id.

Hi Marcelo B., I noticed your profile and would like to offer you my project. I saw you have already done a few integrations for Linphone with asterisk and moved away from Flexisip. I am looking to create a very stable system and dont want to waste time using the wrong software and then having to do it again. Let me know a good time to talk. my whatsup [Removed by Freelancer.com Admin]

we are looking to build system where we can generate script based calls which is going loop back to our server . also we need to maintain ASR and ACD as per predefind critrea

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DynDNS address of the head office, as head office IP address is dynamic.

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DyDNS address of the head office, as head office IP address is dynamic. Time to finish project : 1 day, 2 max

I need predictive dialer and reporting software for our issabel pbx server. Asterisk verison 16.16.1 1- Predictive dialer and reports. 2- After the automatic dialer, the desired announcement will be played, dtmf dialing will be recorded and reported. 3- Search will be made according to queue availability. 4- The number of calls to be made at the same time and the number of redial calls can be entered in the subscriber or queue information to be transferred if the number of calls is answered. 5- Campaign can be created. 6- The dialer will automatically call a list of customers and when they pick up, it will route the call to an available agent. 7- It will be web-based and needs to be written with nodejs react.

...See other light shows customized by random people: You'll be receiving between 5-15$ per light show depending on length and level of detail. A normal one will be 8-10$. The starting compensation for the first 6 songs is 100 $ A customized LightShow is NOT just to apply random lighting onto a regular song. You will be adjusting the sound, light, window movement, and trunk opening/closing to give each light show a fun/unique/jaw-dropping experience. A standard light show is between 60-90 seconds. To begin with, you'll build customized light shows for the following songs: Nirvana - Smells Like Teen Spirit () Eminem - Lose yourself

This project is to setup a web SMS management software and gateway (Gammu, kannel, PlaySMS, Jasmin, etc.) along with an existing Sangoma 7 freePBX system. SMS management and gateway must also support multiple usb modem dongles and sim card so that SMS that comes in from any dongle will be received in the web UI and be able to replied to using a preferred sim card from the USB modems.

I am looking for an open source mobile SIP client. Must work & should be very stable for the following operating systems iOS Android Windows please only BID if you have done SIP & dialler projects before! We need source code and customized UI as well.

Now i used elastix2.4 asterisk 1.8 operation Sangoma A102. Problem is can not fax from E1 PortA to E1 PortB fax always cut of and give line error to distination fax

Looking for a developer for a video call/chat app. With skills in XMPP, WebRTC, SIP

I'm looking for an Asterisk and IVR specialist with voice recognition and "text to speech", who can develop a dynamic IVR with an interface capable of editing prompts, navigation menus and reports. The user will be able to navigate through the IVR via voice recognition or DTMF.

Requirements: At least 5 years of hands-on experience in Real-time communication development (Video/Audio Streaming) Strong OOP skills, fluent with C++ Standard Library. Extensive knowledge and experience with Multithreading and Networking. Deep knowledge and experience with VoIP Stacks, SIP Clients and Servers, RTSP, HTTP Streaming/HLS. VideoAudio codes, packetization, encryption, and transport. Deep knowledge and experience with Video and Audio Streaming. Strong OOD and Design Patterns skills. Skilled and independent architect and developer capable of researching and handling challenging engineering and development tasks.

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...

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

Name of the logo: " Jumping Jumbo". "Need an attractive logo for Toys making company. Has to be colorful and Second J should represent elephant's face or Trunk. " YOU CAN MAKE YOUR OWN DESIGNED LOGO, NO NEED TO FOLLOW THE GUIDELINE IF YOU CAN MAKE IT BETTER " SHOW YOUR BEST CREATIVITY. Nature of Business: Toys, Kids Game, Board Games up to 10 Years old Highly Creative and in 3D and 2D both: Most Imp It should be Original and not copied. Files Needed: png, SVG, jpeg, pdf, and source file. Please comment in the message box for any clarification, I will reply when I ll get time. I will take 10-15 days to finalize the logo and connect for any change or revisions. Please don't message unnecessarily. Whatsapp at Nine zero four one double eight fo...

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

I need to connect SN4130 patton gateway to an Asterisk

Google Assistant webRTC developer This will be 2way call SIP / WebRT job