Asterisk sip gsm jobs
Hi Vinod Kumar I., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
Need someone experienced to create a simple IoT device label. Device is presented on IMG-4469 (1)-min.jpg. We currently have a project of label ( and ), but it doesn't look aesthetically or professional. The label should include: Logo of company QR Code ID code Connector description Device name CE, IP and trash can icons, and also BLE (bluetooth), wifi, LTE-M, GSM and NB-IoT icon
we need an expert in AUTOMATE this is an automation app (if you are not familiar with the app do not Bid ) we need 3 action to be done 1. when app see 0:01 it will send a message to the localhost 2. when the app see gsm call connect it will send message to local host 3 when the android receive incoming GSM call it will close the whatsapp call the budget is 50 ===== TO HELP wuth UI here is
hi , hope all is well. i am looking for someone to help diagnose the connection between my asterisk and sip provider
Need help building PBXact and custom Modules. Should know how to review PBX Status on Asterisk Server and more. We are looking for somone with past experience with Asterisk.
Looking for someone to setup kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamai...kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamailio / OpenSIPs as the gateway and kamailio / OpenSIPs will pass the call to the appropriate ITSP - Setup kamailio / OpenSIPs with local registrations for SIP Clients to handle Video from Door Phones for Remote Clients
we need en expert in Automagic - Android Device Automation to create an 1 step automation using using to trigger an action when a whatsap or GSM call is connected budget is 40
Hello, I need a code, where I wants to send a data to AWS. The data can be any message like "Hello world" and subscribe to the same topic. Please share me price for ssl and non ssl project. Awaiting for response. Thanks
Required Functions (server, port, protocol, username, password) → Boolean Task: • Register at SIP Server • Return true on success • Keep connection alive, reconnect on disconnect • Must support SIP Server, UDP, TCP and TLS connections () → Boolean Task: • Return true if connected to server (to) → Boolean Task: • Initiate a call to a given SIP account • Return true if call is answered (files) → Boolean Task: • Play a list of mp3 files on the SIP call [given as Python list] • Return true if all files are played (char) → Boolean Task: • Send a DTMF sound • Return true if DTMF sound is played () → void Task: • Ends the current call () → Boolean Task: • Stops th...
we need configuration of freepbx virtual machine/asterisk we have about 3 phones to configure there
we need a script or app that will detect if a GSM/Whatspp outgoing call is answered this is 1 action app/script budget 50 dollar
SIP trunk which is located in our Office needs to be connected to IP-PBX (fusion - pbx) which is hosted in Google Cloud Platform through VPN. Also Some parameters have to be passed through X-Headers. Some one who worked previously on scaling IP-PBX preffered
Hi Ayush G., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
...com/spce/). Someone who have enough experience about VOIP, SIP, PBX. You sould be dedicated, serious and working transparent and clean. The sipewise will have to be configured for the moment with one phone provider with one "big" SIP Trunk and multiple channels. In the moment some 30 phone numbers. On sipewise I will generate different SIP Trunk based on the request of the client with one phone number and 2 or 3 or 8 channels. This SIP Trunk "generated" and managed by sipewise will be connected in 90% to an other PBX which the client have or in our data center or in his infrastructure. To secure the SIP Trunk we should think how will be the best solution or to offer to the client based on his possibility up to 3 ways how to connect ...
I am having a hard time integrating asterisk with Odoo. I have set up asterisk in the same server as Odoo, also have installed certificates for webrtc as well as for the odoo portal. I will provide access to my sandbox server.
we need a script that detect if outgoing call GSM/Whatspp is answered (non rooted)
Small multi platform script that: - Registers with a SIP server - Initiates a call - Plays MP3 files - Does DTMF - Ends Call Must work on Windows and Linux "out of the box". All components and source files must be included.
Write a Python script that sends text messages to phones with sip authentication It only authenticates the ip, user and password
Helllo, we are manufacturers of milk and milk products and soon would be launching our own brand for products like- Milk , Paneer, Curd and Desi Ghee. The name which we have decided is White Sip. Need branding and logo assistance
Need to hire someone to connect my Dinstar GSM gateway to X-lite softphone, it is a very micro task. Should hardly take 5 to 10 mins on anydesk.
Looking for Call Generate system for Looping , System Generate calls and it go to supplier and from supplier that call come back to our server if same call come back to our system then only call hold me required duration else disconnect immediately also manipulate ASR and ACD as per our requirement
need some cloud voip serveurs for swithing calls(fusionpbx,freeswith,elastix)
Hi, I wanted an IOT weight machine using a GSM module. The Machine also contains a display and keypad. When people want to check their weight they just want to type their contact no. when standing on the machine and they can visit our website to check their weight. I am building the website I just want only hardware side coding for Arduino.
We have a web app based on Laravel 4 and would like to update it to version 8. Maybe some bug fixing is required too. There are several custom controller, views and models which has been individually developed. It would also be good if you are familiar with Asterisk / VOIP technologies, which is part of the App.
i have found this details at the seller so who can do this project ?[Removed by Freelancer.com] " ATM Skimming without any physical contact. This Skimmer will work without a physical connection to ATM/POS Machine. This product is our best-seller, and is most popular credit card skimming device Worldwide. This is GSM data receiver – skimmer. It receives credit card data from ATM`s and POS terminals. Small size 5×4 Centimeters, antenna length 17 centimeters. With one charging it can work up to 8 hours and the manufactured memory can capture about 100000 credit card data, with antenna it can take data in radius of 50 meters. Device can work from (-25 to +45 degrees). "
Situation We are aiming to dial into a Webex meeting from a SIP device. We have a self-hosted Asterisk server that is connected to our video conference application We are registering this Asterisk SIP account with our video conference application and Dialing into Webex cloud meeting and various telepresence hardware such as CISCO Telepresence, Polycom etc. Complication 1. We are able to successfully connect to telepresence hardware with no problem whatsoever. - PASS 2. We are NOT able to connect to webex cloud meetings (both personal account and organization account). ​We are getting we are getting this 408 request timeout error. - FAIL Screenshots attached for reference Requirement 1. We require you to work with us and help troubleshoot this issue.
I need a Residential floor plan with wall dimensions and elevations. 4 Bedroom, 3.5 Bath, dedicated Office, and a Flex / Bonus Space but no formal dinning. 2 - Story home compatible with SIP systems design installation. 1st Floor: Kitchen, Living room, Master, Master Bath, 1 Bedroom, Office, Utility, 1.5 Bath 2nd Floor: 2 Bedrooms, 1 Bath, Flex / Bonus Space
...Raspberry Pi and have installed (initially) the Asterisk + FreePBX Per this documentation: It got to the point where I was installing the security packages from the command line and it asked if I wanted to overwrite a certain Python file and realized that if I do and it breaks that I need to start over so it'd just be much quicker to work with YOU! So that I can see how to do this properly. Shouldn't be long and we can actually just do it from the web browser (GUI) - in fact we'll have to in order to screen share. I need the RasPBX - Asterisk Dialer to: - Auto-Dialer - Power Dialer - Voicemail Drop (doesn't ring their line, just leaves a voicemail) All of this is actually already baked into Asterisk+FreePBX so it's really just a matter o...
...branding for the chat side of the app. We are now adding audio/video calls including one-to-one and conference calling. As a good reference, you can refer to how WhatsApp and Telegram make one-to-one and group audio/video calls. The one primary thing to note is that we do not use mobile phone numbers for making phone calls so there will not be a dial keypad. We use sip addresses to make calls and each user will be assigned a sip address when they register into the app. The audio/video UI/UX wireframe will include: 1. How to initiate an audio/video phone call. 2. Show one-on-one phone calls in action and be able to switch between audio and video for both the caller and receiver. 3. Include a way to add additional individuals to start a conference ...
I am hiring electrical engineers in following areas: 1) STM32 2) ESP32 3) GPS/GSM 4) Bluetooth / Wireless Mention you areas of expertise in your bid. I would encourage new freelancers to apply.
Greetings, We are looking to get a other few design services from you. Will mention the details below; please let me know an invoice amount for it. 1. A 10 fold (opening out) 50mm*80mm brochure kind, which will detail in t...get a other few design services from you. Will mention the details below; please let me know an invoice amount for it. 1. A 10 fold (opening out) 50mm*80mm brochure kind, which will detail in the 8 oils (will provide the content for it). The first page will be the same photo of the product, and then in next 8; 8 oils with details and at last page details of company. It will be of 30GSM. 2. A 300 GSM 50mm*80mm card which will be like a commitment for goodness card. One side will be written- A COMMITMENT FOR GOODNESS and on second side some matter which we will...
I need someone who knows about asterisk PBX for packet related issues on Asterisk
Im looking for someone can build me a GSM data receiver skimmer credit card for the pos and atm.
1. The Controller will control Streetlight 2. Measurement of Energy consumption 3. Communication through GSM. 4. Data will be sent to server for display and control the controller. 5. Over the air update firmware update.
Hi Folks, We have a standalone asterisk server recently installed on a Ubuntu box. We want to achieve the following in terms of network connection 1. Ethernet interface - This connects to telecom provider for asterisk line and should be used for everything related to SIP 2. Wireless - We want to use this for connecting to the internet Need help from a networking/asterisk developer who can help us split the traffic in the above-mentioned fashion.
...Member's Financial Standing ( Total amount contributed shown next to member's name) 4. Linked to Payment Portal (including Major Five banks in RSA) for online instant payments 5. Automatic Financial Records of all incoming and outgoing transfers with dates and time 6. Ability to automatically send monthly statement of financial activities within the company to member's emails 7. Ability to send GSM text to members every time there is a transaction taking place (Money out only) 8. Ability to upload documents for review by members with limited moderator rights. 9. Have general comment or chat section for members only 10. Have a live chat option for customer care services for public This must be able to be transferred to a host. I would like to know the cost and ho...
Hello, I am looking for a developer who can create a script to sniff phone numbers (sip/pjsip over tls). In order to avoid reading the CDR regularly, I would like a webhook that calls a url. I need date, time, calling phone number, called phone number, transaction id, call duration, status: response codes The documentation for the installation of the script. Environment : Server : PBXware (tls / ssl, pjsip/sip) Smartphone : Communicator / Glocom Linux server (sniffer) : Debian Language: python Library: scapy or other. Transport: tcp / tls
Want to able to transfer a call to another agent/queue using a prefix so we can record a note that is then played to the new agent when they pick up. FreePBX 15 Asterisk 16.13 PHP 5.6.40
A local company designed the hardware and a year later had to re designed it again and I am still waiting for them to finish the firmware the board is a 4 layer the micro controller is a ATXMEGA32A4U-AU the GSM is a MC60 QUECTEL MODULE a 4 button transmitter /recover connects on the board The GPS also check if the vehicle is moving or stationary if the vehicle is stationary a relay on the board is activated that's connected to a op-amp and then to the unit a variable resistance value will then check for changes. all this information will be send to a remote PC. An ignition supply is also required without supplying the wire, when a truck is switched off its voltage will go to 25/25 volt when it start it will go to 28v if that can be used as a switch in the unit if not i would ...
Thanks for reading, we have a problem to solve as below: 1. We are running Asterisk IP PBX soiftware on MT7628 Cpu running OpenWRT 2. For audio we are using PulseAudio and we have some problems: 2a - The echo cancel algorithm for the on-board speakerphone gives echo to the distant end, i.e. the echo cancel is not fantastic and needs looking at 2b - We are struggling to play recorded audio over our speaker 2c - The PulseAudio always requires a second reboot to work properly. If you are an engineer with solid experience in this area we would appreciate assistance on solving these items.
I am researching community cellular networks and need an installation of OpenBTS gsm system that works with software defined radio. Should work in India. Skills Required Telecommunications Engineer, C Programming, Computer Security, Embedded Software, Engineering, Linux, MySQL, PHP, Software Architecture, Software Testing, Web Security, Python Script, MATLAB, ASTERIX Hardware, Raspberry Pi
...connected a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the ext...
Hello, I am running a Small call center with Freepbx 14, needs to develop a customer satisfaction survey application for incoming queue calls. Call landed to the queue > agent answer > agent completed the call > should go to satisfaction survey automatically or agent can transfer the caller to satisfaction survey. Custom IVR should be able to play to the caller, should have a dashboard to view the reports. Reports needs to be contain following Date and Time / Number / Agent's Extension / Score needs to be able to run reports monthly / weekly / daily and custom date range.
We would like someone to organize various events at our retail location on weekends.
Hi Josh, I'm a VoIP communications engineer and developer that can add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint as you want.I have 3 locations listed in my file. 2 are meant to load balance eachother. At the moment it works perfectly as the logs show the calls doing a round robin between the 2 locations. We use 2 sip providers. 1 gives us 2 trunks to load balance and the 2nd we use just for failover in case the first provider goes down or is unable to route a call for any reason. Here is my file: The config file I am using was created by someone else for a different system we use to use. # $Id: ,v 1.1 2004/08/10 16:51:36 dcm Exp $ # sample config file for dispatcher module debug=2 # debug level (cmd line:
Hi Josh V., Hi Josh,.... I am an expert VoIP developer. I will add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint.I have solution for your problem. public virtual HttpResponseMessage Get([FromUri]TwilioRequest request) { TwilioResponse tr = new TwilioResponse(); if ( == "Hello World") ("Hello back!"); else ("Text 'Hello World' for a friendly message."); return (, ); } SMS API enables you to send and receive text messages to and from users worldwide, using our REST APIs. Programmatically send and receive high volumes of SMS globally. Send SMS with low latency and high delivery rates. Receive SMS using local numbers. Scale your applications with familiar web technologies. Pay only for
Hi, can someone do the following: * Remove Logo top left of each page and replace with logo on the business card that blue loop with business name - Resource Appraisals * Change the top of the color of the bar to match...Replace fax line (800) 621-7070 with (866) 906-9326 * Edit the copyright information at the bottom of each page from 2010 to 2002 * Remove the line after All rights reserved with Resource Appraisals and the logo are trademarks of Resource Tech Appraisers Then make the document into a fillable form for Mac / PC, Internet. It will be sent to people to fill it all out and submit it to send back. Red Asterisk are required fields - All fields should support Alphanumeric characters. Under Additional equipment and Notes should support max characters for a long message...
Hell...a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the exter...
