Program requiring linked lists, dynamic array stacks and queues knowledge(advanced knowledge).
I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration
Hi we are looking for a talented mobile game developer to build a mobile game for our prize based app. The app will allow the users to play the in app game to win the prize. We need a developer that’s familiar with monetization using admob, appodeal, mopub ext. I will share the app prototype in chat.
Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (Dinstar gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted ...
We are trying to build a new Asterisk server using PJSIP instead of the older SIP module and need some help to make it work. The setup is as follows: Asterisk certified/16.8-cert3 running on CentOS 8.2 with a public IP address. Phones in the office need to connect to the Asterisk server, they are on a private LAN (192.168.1.x/24) and access the Internet via a NAT router. Two types of phones: Aastra 6757i and Grandstream N300. The server also needs to send and accept calls over a SIP trunk to another Asterisk server also on a public IP (PSTN Provider). You will need to be an expert with Asterisk and PJSIP and work with us to get this working and explain what you did - you can have SSH access to the Asterisk server if required. For the right pe...
The zip file is a conventional zip file - not an Eclipse package archive. Create your own Eclipse project, unzip this file and add the .java files to your src folder and the data files to your project folder.
...Call/External Call Ringtones (Included) NTP Synchronized Date, Time, Events (Included) Individual Named, Numbered Extensions (Included) Directory of placed, received, missed calls. (Included) Ad-Hoc Conference from VoIP Phones to Any Phone (Included) Inbound/Outbound Calls From All Phones and Extensions (Included) SIP Trunking (Included) Auto Attendant (Included) 8 Individual voicemail boxes responding to ext. Plus 8 virtual voicemail box (Included) All call report capability to include number and time date stamp, time on call (Included) Security: Auto Secured Firewall (Included) Voice, Data, and Management are in separate VLANS/Subnets. (Included) System is hidden on the internet as it cannot be pinged or port scanned from outside. (Included) Hosts on the internet cannot talk t...
...es/BMO/(124): FreePBXmodulesCertman->__construct(Object(FreePBX)) #4 /var/www/html/admin/libraries/BMO/(37): FreePBXSelf_Helper->autoLoad('Certman') #5 /var/www/html/admin/libraries/BMO/(298): FreePBXSelf_Helper->__get('Certman') #6 /var/www/html/admin/libraries/BMO/(39): FreePBXHooks->preloadBMOModules() #7 /var/lib/asterisk/bin/retrieve_conf(81): FreePBXHooks->updateBMOHooks() #8 {main}...
...checking target system type... x86_64-pc-linux-gnu checking for PHP prefix... /opt/cpanel/ea-php74/root/usr checking for PHP includes... -I/opt/cpanel/ea-php74/root/usr/include/php -I/opt/cpanel/ea-php74/root /usr/include/php/main -I/opt/cpanel/ea-php74/root/usr/include/php/TSRM -I/opt/cpanel/ea-php74/root/u sr/include/php/Zend -I/opt/cpanel/ea-php74/root/usr/include/php/ext -I/opt/cpanel/ea-php74/root/usr/ include/php/ext/date/lib checking for PHP extension directory... /opt/cpanel/ea-php74/root/usr/lib64/php/modules checking for PHP installed headers prefix... /opt/cpanel/ea-php74/root/usr/include/php checking if debug is enabled... no checking if zts is enabled... no checking for gawk... gawk checking whether to enable internationalization support... yes, shared checkin...
Hello, We need to create extensions and edit some params through an api. The parameters we need to be able to edit/create/delete are; -Extension number -Extension alias (display name) -Direct DID -Outbound DID -Inbound Direct DID -Password -Follow me settings (initial ring time, followme list, ring time, confirm calls) Our Asterisk is an Issabel distribution.
We need to develop a softphone application to work with Asterisk freepbx.
we already have a chrome ext that is able to scrap data from a social networks (LinkedIn) and send its back to our API. we want to add more social networks to the chrome ext. - Facebook - Instagram - twitter We are looking for a developer who can accomplice those.
We have a bug in our implementation of Kamailio + Asterisk + Mysql Proyect, we don't know why, but the Asterisk crashes with Memory problem.
We have purchased a codecanyou Saas Laravel ready-made script , and we found it not properly functional, we looking to make it work and customize some parts, the following are the tasks: 1. Contact us - Can't send emails, Solution to setup Laravel supervisor: 2. Upload profile not working, we are using Plesk and it works as shared hosting. 3. Adding terms and conditions page. 4. Customize Subscription Page + update and integrate a payment gateway, it also should accept free packages. 5. Clear all demo data 6. Ticket is not working 7. Coupons not working 8. Notification not working 9. Team page not working
We have purchased a codecanyou Saas Laravel ready-made script , and we found it not properly functional, we looking to make it work and customize some parts, the following are the tasks: 1. Contact us - Can't send emails, Solution to setup Laravel supervisor: 2. Upload profile not working, we are using Plesk and it works as shared hosting. 3. Adding terms and conditions page. 4. Customize Subscription Page + update and integrate a payment gateway, it also should accept free packages. 5. Clear all demo data 6. Ticket is not working 7. Coupons not working 8. Notification not working 9. Team page not working
We want to have sip trunking between Avaya and asterisk pbx.. If anyone have experience and stays in Jeddah let us knw
...functionality to import XLSX files to update specific class. fields in the class are in classification stores. Thanks, Ofer Bundle that Imports Excel On Pimcore Admin Backend UI 1. Pimcore version 6.4.x 2. The software must be written as a PHP Symfony Pimcore Bundle installed on Pimcore Admin backend UI. 3. The user interface should be based upon ext-js. 4. The bundle should let the user on Pimcore Admin load Microsoft Excel (xlsx) file to update the fields values of a class. 5. The excel will be prepared by the user. 6. There is only one class that should be updated. (only update – no insert). 7. The class contains fields in Classification Store. ...
...functionality to import XLSX files to update specific class. fields in the class are in classification stores. Thanks, Ofer Bundle that Imports Excel On Pimcore Admin Backend UI 1. Pimcore version 6.4.x 2. The software must be written as a PHP Symfony Pimcore Bundle installed on Pimcore Admin backend UI. 3. The user interface should be based upon ext-js. 4. The bundle should let the user on Pimcore Admin load Microsoft Excel (xlsx) file to update the fields values of a class. 5. The excel will be prepared by the user. 6. There is only one class that should be updated. (only update – no insert). 7. The class contains fields in Classification Store. ...
Freelancer willl provide expert advice for GOIP equipment and settings. Please let me know about your experience.
We need additional developers to develop a pwa based on the TYPO3 CMS with an individual domain model. Basis TYPO3, nuxt-typo3, TYPO3 headless ext
Hola Felipe V., vi en un mensaje que publicaste que sabes de asterisk y grandstream. Tengo un problema y no encuentro la forma de solucionarlo. Te dejo un link con mi problema, si consideras que puedes solucionarlo me gustaria que me contactes y me digas cuanto me cobrarias. Gracias
] WARNING[21199][C-00000007]: func_base64.c:79 base64_helper: Syntax: BASE64_DECODE(<data>) - missing argument! this is what i get when i am supposed to receive a ussd asterisk 16
My name is IEVGENII DMYTRENKO. I am an owner of used car dealership in Ottawa. Looking for a professional to built a web-site for me. Web-site has to have Buttons for: inventory, finance, about us, contact form, message pop up, payment calculator, auto finance form ext.
Custom Themes for Vicidial and Call flow changes for asterisk for Inbound Call Centre SIP based Outbound Call Centre Sip based IVR UI based Predictive Dialer Pls share the demo of your work on Vici Dial
I have a server with all my asterisk files that I need someone to fix up and make admin panel for me
APLICACIÓN SIP CALLING QUE REEMPLAZA LOS TELÉFONOS DE LA HABITACIÓN DEL HOTEL Y AÑADE NUEVAS FUNCIONES. El objetivo del proy...recepción, el spa y más. Al estar diseñado como una aplicación para un dispositivo móvil, el producto aseguraría que los huéspedes tengan la capacidad de realizar y recibir llamadas para servicios de hotel en cualquier lugar de las instalaciones del hotel. Ya disponemos de servidor SIP asterisk con API específicas para las funciones necesarias, tales como alta de cliente, alta de cuenta sip, activar/desactivar cliente. La cuenta SIP creada será la que se use para registrarse internamente el sistema en el servidor asterisk. Es recomendable nos envíen de...
...a POC of using Asterisk for our phone support. I am an experienced software developer and an engineering manager, but I have no experience in Asterisk or PBX. This project is to run Asterisk on a new Ubuntu/20 server running on AWS, listen on SIP calls and handle a few basic operations: 1. Voice greeting, coming from mp3 or other audio file. 2. Connectivity to Google Voice Recognition, listening for next commands 3. Logging the recognized voice in text and in audio (mp3 or other audio file) 4. Responding with another greeting from mp3 files, depending on recognized voice (3 options: option A option B or unrecognized) Also, we own 2 VOIP DIDs in , which has SIP connectivity and DID POP. We would like to connect those, so that calls to these VOIP will reach the As...
I am looking for a python programmer to ...documentation. Other than the example provided in the code, I'm not sure how other Classes should even be used. The assertion tests will (in part) provide an example of how each Class and Method are used. I would like assertion test for the module in You can find sample .ini files here: My primary interest is how to use append, insert, pop, and remove in the Category Class. I assume the module is capable of modifying sections and key/value pairs. Using coverage and pytest, your objective is 100% coverage, pass or fail for the module.
We currently are using asterisk version 11.22.0 We have BLF currently working Asterisk Feature Busy Lamp Field (BLF) so it is working but we have to create an BLF subscriber extension like this [BLF] exten => 1000,hint,SIP/${EXTEN} exten => 5000,hint,SIP/${EXTEN} exten => 5001,hint,SIP/${EXTEN} exten => 998899,hint,SIP/${EXTEN} exten => Prox1,hint,SIP/${EXTEN} But this is working. when the command asterisk -rx 'core show hints' run it shows hints for each extension This works but we want to create a simpler BLF BLF subscriber extension like this [BLF] exten => _XXXXX,hint,SIP/${EXTEN} But this is not working. when the command asterisk -rx 'core show hints' run it does not show hints for each extension We need woul...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - U...
Looking for someone to develop an IVR survey based on Asterisk pbx software, need guidance from scratch
The goal of this project is to learn about process scheduling inside an operating system. You will work on the specified scheduling algorithm and simulate its performance. In this project, you will simulate the process scheduling part of an operating system. You will implement time-based scheduling, ignoring almost every other aspect of the OS. Please use message queues for synchronization. The operating system simulator, or OSS, will be your main program and serve as the master process. You will start the simulator (call the executable oss) as one main process who will fork multiple children at random times. The randomness will be simulated by a logical clock that will also be updated by oss. In the beginning, oss will allocate shared memory for system data structures, including ...
Need someone to set up an asterix server for us; this would include connecting to the outbound SIP provider (twillo) setting up the GUI, some pilot groups, conferencing groups, automatic call recording, call transfer, sms response to incoming calls that are unanswered, a Grandstream DP750 endpoint, some soft phones, and probably a few other things on a Debian install this would also include some asterix education along the way .. I can provide SSH access for the system at a clean install point
Hello, I need to to know complete potential usage of the asterisk as well as setup asterisk Thanks
I need to configure Vtiger Asterisk Connector to do click to call (outbound and inbound calls). I have my Vtiger installed and my own Pbx
looking for some one to implement an app for push to talk PTT with asterisk as a back end .. The android app will use the functions and features of Asterisk to create push to talk functionality This message to call candidates : If you have no asterisk experience .. it will be difficult for you to deliver the work to be done . we are using a system open source system called Asterisk for SIP communications .. and we want to use the same system as a backend and build an android app as a push to talk app
i need to join goip and skyline with asterisk for make calls and in same time join goip with api to build new smart interface and send ussd and sms and do recharge sims etc in same time need asterisk unser some dail plan delay call to send to goip thx
I am searching for an expert who can help me building cluster and load balancer with Asterisk
Hello I have an asterisk pbx that works behind firewall and the way it works is with a router that allows IP traffic destined for SIP communications from internet on dedicated WAN port to be directed with no inspection to and internal IP of the asterisk PBX, also all internal lan trafic to asterisk PBX is thru the internal IP address. In the router all outbound SIP traffic is redirected thru the same inbound WAN connection solely for SIP communication. This works fine with and Linksys RVS042 but I need to change to a mikrotik RB2011iLS-IN and what I would need are instructions on the changes to be made. No external access will be provided, only instructions and network topology can be given.
I need an Asterisk Expert for certain task. Let me know if you have the ability and expertise on Asterisk. Thanks!
We're developing an app which has an unepected need for Asterisk integration. We need 1 extra developer to help setup asterisk with SIP trunk to take data from a python program, and make a phone call with that data. The call will need SpeechRecognition implmeneted, and play DTMF tones.
Heaps and Priority Queues - in Python Develop a heap data structure using a linked structure (Nodes and Pointers) The heap must support add and remove from the heap All operations most abide by the rules that govern a heap (see lecture slides for reference) Once you have your heap structure created, next you must use it as a backing structure to a priority queue. Develop a priority queue data structure that is backed by a heap (linked structure NOT A LIST) Implement the normal methods that accompany a priority queue structure Enqueue, dequeue, and peek by priority not position Also length and whether or not the structure is empty (is_empty) Perform the following operations to showcase your working structure Enqueue the following items: 4, 7, 5, 11, 8, 6, 9 Dequeue 3 items by priori...
we are looking for a consultant to develop a microsoft teams add-in that consists of a webrtc application for voip pbx (asterisk etc..)
Hello .I need videos on sewing If you are perfects on sewing .broderie or crochet ..ext or other hobbies Make a video tut sewing something like dress or other The video must be clear clean and food quality Must be +10min I pay 5 $for each video
This is a program that will simulate wrestling matches between teams and bouts between wrestlers. It will generate the wrestlers/teams at build time and place wrestlers into an array of priority queues using pointers depending on their assigned weight class and ability score (AS). The simulation will include 32 different teams (100 to 3200), where 8 of which will be in a single league (4 leagues). Each season will have a number of dual meets, team vs team, and four tournaments over the weekend (first 3 are open for 8 teams, and last one is for the 8 league schools). - Match is Team vs Team - Tournament is 8 teams competing in a bracket style more details in the inbox budget is $50
This is a program that will simulate wrestling matches between teams and bouts between wrestlers. It will generate the wrestlers/teams at build time and place wrestlers into an array of priority queues using pointers depending on their assigned weight class and ability score (AS). The simulation will include 32 different teams (100 to 3200), where 8 of which will be in a single league (4 leagues). Each season will have a number of dual meets, team vs team, and four tournaments over the weekend (first 3 are open for 8 teams, and last one is for the 8 league schools). - Match is Team vs Team - Tournament is 8 teams competing in a bracket style more details in the inbox
This is a program that will simulate wrestling matches between teams and bouts between wrestlers. It will generate the wrestlers/teams at build time and place wrestlers into an array of priority queues using pointers depending on their assigned weight class and ability score (AS). The simulation will include 32 different teams (100 to 3200), where 8 of which will be in a single league (4 leagues). Each season will have a number of dual meets, team vs team, and four tournaments over the weekend (first 3 are open for 8 teams, and last one is for the 8 league schools). - Match is Team vs Team - Tournament is 8 teams competing in a bracket style Each team will consist of 13 weight classes, where each weight class is it's own priority queue of wrestler pointers. The number of wre...
