Asterisk definity sip trunktrabajos

Hello, I would like you Caller Id asterisk sip that can be used anywhere in the world. Thank you

We are looking for an asterisk programmer to create a module to manage automatic outgoing calls campaigns using TEXT to SPEECH automation in ISSABEL or FreePBX. The manager user must upload a CSV file that constains all the information of each customer to be contacted and when the call is answer the system has to reproduce the audio file in spanish with the specific information for this customer, it is not reproduce the same audio to all users. The module must let the user to generate all the reports to know the results of the campaign (Contacted, not contacted and time of audio reproduction, etc)

We are currently working on VOIP project, the GUI part of our project is completed. We want DID Number, Ring Group, Call Queue are automatically implement from our GUI to our asterisk backend.

I use 3CX startup. I want to add a sip trunk for calls in Belgium. I have problems with provisioning my Yealink W76P (W70B & W56H) and use it as a router phone.

We updated asterisk to latest version. Now we are facing the issue, that in some of our scripts the output of the AMI Calls are not parsed correctly. We need you to fix it. Please read the article (this describes our problem)

We need to finalize the goautodial v4 configuration on a cloud server: 1- Finalize the configuration of the sip trunk 2-Deploy a campaign (test) 3-Configure incoming calls

This project is about customising Vicidial & Asterisk applications. + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

IT Service im Raum Hessen/Bergstrasse Virtualisierung auf Basis von Proxmox un VMWare,Firewall Konfiguration mit OPN Sense,Windows Server Umgebung, Docker und Kubernetes, VoIP mit Asterisk und Freeswitch

...completion the coding. Looking for a Senior Java Developer to finish coding XanticTek TASS application. Must have skills: • More than 5 years of professional Java 8 (or higher) development experience • Understanding of maven multi-level projects • Knowledge of Java Swing (for the UI) • Writing SQL queries for Postgres DB • Good level of Linux knows how Nice to have: • Deeper knowledge of VoIP and SIP protocols If you don't meet the “Must have skills” and you don't have the curiosity to learn “Nice to have” there is not making any sense to open discussion with us. Category: Software Domain: VoIP (voice over IP) Function: AntiSpam System for VoIP & SMS traffic Documentation: all Features are extremely well pr...

We have a problem with a Cisco ATA-192 that will not register with our Asterisk server. With tcpdump we can see REGISTER attempts from the ATA and a 401 response being sent back, but nothing further. We suspect it may be a NAT issue. The remote end is behind a broadband router and is NAT'd. We need a network expert that also understands Asterisk and SIP. Also, ideally how to configure a Cisco ATA-192.

Two images blended together in a clean style. Logo is to be a dark grey on a white/clear background. The eye is to replace the arrow head at the top of the tree symbol. At the bottom of the tree the letters "SFAHQ" are to remain but the tree trunk is to be removed from the bottom section so the letters do not clash with it. THE IMAGES ARE PROVIDED AS A REFERENCE ONLY - DO NOT TRY AND CUT PASTE THESE ACTUAL ONES. This project is someone who can graphically design, not just cut and paste.

Hello I am looking for designer to perform a SIP design home about 1500 sqft. on a down slop hill of 40%. 24x56 ft. athe picture will give the idea.

...to replace one of the 3rd solutions we're currently using called - Zoiper (softphone) We're currently seeking a service provider with previous experience in Flutter or React Native to build an application for like IOS & Android. The candidate should have previous experience in building communication applications and also have a strong understanding of how VoIP, PBX, Asterisk work & interact behind the scenes. The app will be integrated with the open-source system, Asterix, and serve as an interface for issabel. The tools that we currently user are: If you have experienced in building communication apps and understand Asterix, VoIP, FreePBX & Flutter, please reach out

I would like to develop a SIP/VOIP application for mobile and desktop which will be used in fusion pbx

As per our discussion over the chat conversation: 1. Configure Asterisk to perform a cURL request to a PHP Endpoint posting Caller ID Number 2. Display upcoming call informing from MySQL through a Popup using AJAX and Javascript to a PHP-driven page

I need to monitor my siip server. The correct freelancer will suggest which app/service is best to use for this.

Hello, we are a hosted voip service provider and until now we have been using FreePBX and Asterisk, we are looking at moving away from FreePBX. We are open to either Asterisk or FreeSwitch, whichever meets our requirements listed below: Good day, We are a VoIP Solutions Provider that is currently looking to move away from our FreePBX systems we host for clients to a Class 5 Softswitch/PBX. I have compiled a list of features we want but don’t currently have with FreePBX and then the top features of FreePBX that are an absolute must have in this development. This would need to be a linux based platform capable on running on multiple dedicated servers with iSCSI storage. Ideally, we would like the platform built on a AlmaLinux or Rocky Linux. We would want the abilit...

We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams

We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams The goal of this project is: Configure an Opensips server with Opensips Control Panel that: - Connect to asterisk PBX server / Fusion PBX - Connect to Microsoft Teams - Let users from MS teams call users on Asterisk / Fusion Extensions and external calls though this PBX - Let users from Asterisk/Fusion call users on MS Teams

A copy of the poster is attached, 5 poster a week $20 per poster. Please i am not asking for too much stories. This are all the poster i need. Exotic Painting Paint & sip Booking; minimum 10. group 10% voucher food promotional ladies night @hollywood entertainment. Lunch special African food birthday parties. Book ur event. Veunue Hire Spicy Saturday event. Every Saturday Every Friday Karaoke like us IG : FB website. @myplacebarperth

we have FreePBX server and we need to configure the IVR dial Plan with setup SIP trunk inbound and out bound

Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.

Hello. I have 125 Google sheets of Multiple-Choice Question Data. These are all formatted identically and an asterisk is used to identify the correct answer. An example of the first sheet of data can be seen here. To import all of these quizzes into my website I need to reformat each sheet automatically to match the new template. The biggest change here is that an asterisk is no longer used to identify the correct answer and instead it is a numerical value of 1-4 which indicates if the correct answer was A,B,C, or D. Ideally, I would like to be able to copy the original sheet of data into a book, run a script and end up with the data formatted correctly for export. An example of the new

I have a white-labeled Linphone application for both iOS and Android completed and use Asterisk servers. Push notification for incoming calls is causing me challenges. I'm looking for someone to hire to: Walk me through setting up my Apple developer account and Firebase to send push notifications to my Linphone build. Walk me through associated modifications to Linphone build (integrate Google plist, etc, whatever needs doing). Configure kamailio (preferred) or flexisip (acceptable) to proxy between my Asterisk systems and Linphone on client mobile devices to handle push. We'll use a fresh Debian 11 install that I will provide you credentials for. If this is something that you can take on, please provide a quotation.

I have a white-labeled Linphone application for both iOS and Android completed and Asterisk servers. Push notification is causing me challenges. I'm hoping I can hire you to: Walk me through setting up my Apple developer account and Firebase. Walk me through associated modifications to linphone build. Configure flexisip proxy to handle push on a fresh Debian install that I will provide you credentials for. If this is something that you can take on, please provide a quotation.

To invite everyone around Little Alden and Spring Lake Sunday December 11th, between 1pm and 4 pm At Cindy and Rod's, 3296 N Little Alden Lake Road, 218-391-5815 We'll have Fires going...we'll have Hot Chocolate to sip on and all the ingredients for you to make Smores ! Bring any Beverages that you'd like !! Sit around the Fires, get Cozy in the Bunk House or Relax in the Cabin Wear Some that is all Christmas...A Hat, a Sweater, some Socks...We'll try to get some good Pics to put together in a collage !! We hope to see you Sunday ! We just need a basic flyer to distribute around the lakes and to email. Use some Christmas colors...make it fun... I am looking for an inexpensive flyer that is professionally done..." not in my handwritng " Than...

...need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matter the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, that triggers successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to the WhatsApp gateway 2) WhatsApp gatew...

requiero un manual o tutorial para implementar seguridad en el servidor asterisk, ya sea con fail2ban, iptables. para mitigar los regitros sip, ssh y los ataques de denegacion de servicios. Acepto cualquier sugerencia para mejorar dicha seguridad. I require a manual or tutorial to implement security in the asterisk server, either with fail2ban, iptables. to reduce sip, ssh logs and denial of service attacks. I accept any suggestion to improve said security

I need to configure VPN on YEALINK T29G phone. I have a soft-ether OpenVPN server running but I am struggling to configure as CLIENT my YEALINK T29G Sip Phone.

1. Vendor will ...Virtual CUCM (Unified Call Manager) on the Virtual Machine, The VM will be provided by the Customer on their ESXi license 5. Perform Standard & Advanced Configuration of Cisco UCM 6. Implementation of 30 qty. IP Phone 7. Rack Stack, & Power of New Voice Gateway Routers ISR4331-V/K9 (Qty. 1) 8. Initialize Voice Gateway Router • Perform staging & Software Upgrades • Configure OOB management 9. SIP trunk established between Cisco UCM & Voice Gateway Router 10. T1/E1 termination and its configuration for outbound call 11. Configuration of logical partitioning in CUCM 12. A-Flex-3 Smart license allocation 13. Overall IP Telephony system testing & go-live of the deployed solution with Customer 14. Submission of Implementation do...

WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk

We installed goautodial v4.0 from iso Kamailio running HTTPD OK SSL certificate OK RTPENGINE Ok Our main issue is the following: 1) Agent need to press (Login to dialer 2 times) 2) Can't register GoIP gateway (SIP Trunk) 3) Can't hear any voice. Only Goautodial V4.0 specialist is required...!!

Im looking for someone to build be a self hosted voip system We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc'

Hi, We need someone to design a brochure for us. The brochure will be sent to companies to introduce our paint and sip events. Most of the photos will be provided + a sample. However, we need a creative person who can design something unique, very clean design and professional look. Photos will be provided, the person might need to write some parts by him/herself. We need someone who have a background in marketing.

webRTC SIP / Chat / video / conferencing / Electron / typescript Android / IOS app need to develop a app to work in electron / android /ios / angularjs

I'm looking for someone to develop me an Android app that acts as a ""SIP server / mobile gateway" and makes the cellular line of the smartphone usable with a SIP client. So it is also a kind of VOIP mobile gateway. It may be possible to port the free IP telephony software to use the cell phone as a cellular gateway. However, it is important that the APP serves both as a gateway and as a SIP server, so you can log on to the app with any VOIP client to make calls via the smartphone's cellular network using SIP protocol (IP telephony). Important: The app should run on any modern Android smartphone running Android 10 or later.

Hello We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc' The end goal is to be able to start selling voip services. We prefer a person who did something like that before. I will be happy to answer questions.

Hi - we have an asterisk box V16 running on Centos 7. We're having ongoing issues with CPU spiking with as little as 3-4 calls, reboot of the server temporarily resolves the issue but we need to resolve the root cause.

Make PWA SIP-client be able to receive calls at lock screen mode ( iOS/Android smartphones). I know the current problems with push on both platforms :-) But Im looking for solve it. And want to pay for solve.

Hi, I need someone who has an excellent experience in the FREE Open-Source SIP SBCs to suggest the vendor we will install and help me to install it Kindly don't bid if you don't have the experience that is needed, Thanks!

Asterisk Server Script not working I will send details in chat

I have two drawings. One of the tree . Complete with leaves and another of the leaf only in greater detail that shows how I want them to appear in the drawing. I need the file to closely match the drawing . I want to be able to use the pic with a background color and the color to be able to pass thru the gaps I left in the tree trunk and also the leaves. The gaps in the tree trunk need to be evenly spaced. I don't want any borders or outlines on the individual segments of the tree. I want to be able to color each individual segment with a different color if I wish. I would like the drawing to be in PNG format.

We have a concept, colours and design just need someone to pull it together and present digital formats of logos. Colours are #2c4c3b & #2c4c3b We like the design of the tree with the dark green background but want to bring in the elements of our old logo. If you look at the old Logo, the leaves are actually small butterfly's (We want to retain this) and the centre of the tree trunk is actually ida and pingala. The Two intertwining parts of the tree through the middle are what we want to retain. Finally we would like the feeling of an aurora around the logo

We need to modify just one function in open-source software that is connected to our Asterisk (FreePBX) server. We know where exactly this change should be applied. So it would be very straightforward for a java experienced developer. Asterisk, AGI, AMI, Java-asterisk lib, Java, J2EE, PHP, Python

...Ambient music + creature noises 10. Effective draw distance for a multiplayer web browser game 11. Proximity voice chat (based on player vicinity to sound source or other player) 12. Stream a real-time video feed (2D Sprite) of Maghmul into the Unity environment* 13. Movement mechanics (WASD/Gamepad + Jump) for players and real-time videostream* 14. WebGL Deployment and testing * Items with an asterisk are already in progress or completed Kindly refer to the videos linked at the bottom of this document for more information on the status of the project Ideal Candidate 1. Would be able to provide fixed-bid quote or estimate for the items listed in Phase 1 following a 4hr introduction to the code base. 2. Have 3-5 years full-time experience developing games or experiences in Unity ...

Create custom module in prefex crm for integration with asterisk with following features: - create/edit extensions - create/edit trunks - create/edit outbound routes - create/edit inbound routes - create/edit IVR menu - calls report and recording - popup up screen when incoming calls with customer details or add new option. - click to call from crm. - conversation history in customer details. Thanks

Project Requirement: 1) Design one Linux Script with netcat (nc) or standard C program for sending the "Caller ID" from my Asterisk IP-PBX to the WordPress Web server. The Linux Script Command options will look like "send_hostip_callerid". 2) The WebPress Web server need one simple Server Daemon to receive the "Caller ID" sent from my Asterisk IP-PBX by "send_hostip_callerid". The Server Daemon on WordPress Web Server need to support multiple concurrent incoming connections or you can use the CGI program of the WordPress Web Server to handle this job. After this daemon or CGI program has received the "Caller ID" from "send_hostip_callerid" on Asterisk PBX, it must save the "Caller ID" and current ...

I want my ASTPP Server to make outbound and Inbound calls from the Sip trunk routed from its second ethernet interface which it can primarily see when you ping directly. but says Gateway Ping Failed on Fs_cli

+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

We have sip truck placed and need to configure call inbound, outbond and ivr with freepbx