We are planning to build a Automated IVR system, we would like to integrate the below systems 1- Implement Media Server using UniMRCP 2- Integrate / Enable ASR (Google Speech ) with UniMRCP 3- Integrate UniMRCP with our Conversational AI Platform 4- Integrate Asterisk / UCCE(Unified Cisco Communication Enterprise) for a end to end voice journey
we have few Asterisk pbx systems. We have an issue where if we forward a call to an external number it works only when the caller is called with CLI enabled. if someone calls as anonymous call fails. Our upstream carrier has below mention requirements. CLIs will be accepted in the Incoming INVITE SIP headers in the following priority order: 1. “P-Asserted-Identity” header 2. “From” header If a private/hidden CLI is required, then the P-Asserted-Id Header must be included and contain a valid Australian or International CLI number with the “Privacy: id” header also included. Headers NOT Supported for CLI delivery: • Contact header • Diversion header • Remote-Party-Id header Calls where the “P-Asserted-Identity&rdq...
We would like to integrate an third party software (SIP call) into our software, so that we can apply the SIP video call functions in our software. Our software is developed in Cordova and the third party is in Native so we need to convert their functions as cordova plugins for both IOS/Android. The third party software is Comelit intercome software, it can be found from IOS/Android store. Condo Intercom Working Scenario: There will be a call panel outside lobby door, when Visitor / Deliverer reach condo, they press the unit number of resident. Then resident phone app will receive notification of calling in request. Then resident open app (or agree to call) to talk to visitor via call panel. There will be a camera on call panel so that resident can see video stream from app....
I am looking for someone to setup and configure Kamailio or Janus ( Preferably Janus) as a WebRTC to SIP Gateway in from of multiple Asterisk server. Janus / Kamailio should be able to load share to multiple Asterisk servers based on Caller Prefix.
* Merge 2 spreadsheets into one sheet. Serial number in column A used for matching. Are you a bot? If not, put an asterisk (*) in front of your proposal (see how I did it?) Each sheet has 5500 lines.
1. Using issabel call center module while on call with customer need to hold the call, this data need to capture in database. 2. after call finished needs pause option for (ACW) this data also need to capture in database 3. need to resume the call before default ACW settings
Hello we are looking for a specialist to build a wordpress website with all conditions that will be required We will provide the content and design in figma 2 languages website Payment integration - on 3rd part payment website Chat integration - facebook messenger, telegram, viber Control Panel - for changing, adding information to website Website Design Link:
Hello we are looking for a specialist to build a wordpress website with all conditions that will be required We will provide the content and design in figma 2 languages website Payment integration - on 3rd part payment website Chat integration - facebook messenger, telegram, viber Website Design Link:
We are running Asterisk 1.4.24 and we are experiencing random deadlock crashes (segmentation faults w/ core dumps) when using the chanspy feature. Core dumps are available
Hello, We need to adapt an existing asterisk context to handle failover. We use freepbx.
Windows based asterisk voip to gsm gateway using real mobile phones connected to the USB ports which are recognized as extensions.
...(handling calls). ASTPP2 needs to be connected to ASTPP1 DB, so that in ASTPP1 GUI we see also ASTPP2. Users from ASTPP1, must be able to call to user connected on ASTPP2. In ASTPP1 GUI, we must see the customers registered on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only). The done job must be documented from the beginning, very detailed, how the ASTPP's were connected, after installation, what commands are used to achieve the desired result with proofs of successful tests. The doc must be very detailed and clear, as a reference freelancer takes...
We want to setup telegram to sip with this below link instructions - Raspberry pi product link - Is it possible for you to create this setup and make workable, if so let me know how much time you need for this project and what is the best possible rate that you can offer us ?
Developer for creating chrome plugins for popup windows. This pop-up app should connect to Asterisk and have the following features. 1. Connecting to Asterisk via the ami protocol 2. When you receive an incoming call, you should a) a pop-up window will appear. The caller's number should be displayed in this pop-up window. b) open the URL to which the caller number should be added by the template. 3. Make outgoing calls, by clicking on the number or from the call history
I am looking for rebranding & skinning Jitsi open source software including , IOS and Android Apps and web application. Experience in WebRTC, SIP , Javascript, PHP, Aache, SQL , JAVA , IOS & Android are mandatory. Additional preferable skills include Asterisk , Video Conference , ZRTP. I would need the signed IOS and Android App along with web application and server side application all rebranded. All functionality of Jitsi should be available.
I need Asterisk installed on my remote server please. So the end result is the same as this: I only need outgoing calls, I will sort out my own trunks. I have attempted myself numerous times. I cant get it working. Many issues when I try it is my first time using asterisk and I am fairly new to Linux so very novice here. Thank you, for someone experienced shouldn't take longer than a couple hours.
I need a logo for my business i sell beauty products & Perfumes, i need a simple & elegant logo color PINK
Converting 4-site CCM install to SIP trunks. - Main site has 2 CCM, 2 gateways with 2xPRI - 3 remote sites connected via MPLS have SRST routers and mixture of POTS lines Project requires: - Add Internet to each remote site (customer provisioning) - Add SIP trunks to each site to replace POTS - Provide failover scenarios (fault tolerance features enabled by SIP service provider) - Configure CCM to handle new call routes - Any other changes - As built documentation CCIE/Voice Required
Hello, I need another article translation from English to French. The article deals with SIP so some technical understanding is required. It's a bit longer than 1 page. Are you interested in this job?
Hello Experts, I am interested in developing a project with WhatsApp, basically I need my clients to be able to contact us through wsp and when they leave us a message a chatbot must respond and then they must refer the conversation to any user who is logged into the wsp application, for greater detail the application must do the following: - Fixed num...refer the conversation to any user who is logged into the wsp application, for greater detail the application must do the following: - Fixed number association - Users - Chat Bot - Web Portal - The application must be able to be installed on a Mobile (Android or IOS) - Contact management - Quickly answers - Call button from the web interface (It can be WebRTC. Also, the call should be sent to an Asterisk PBX). - Conversation hist...
need a xamarin andriod voip app using Ozeki VoIP SIP SDK
Hola Francisco M., veo que eres un experto en asterisk y freepbx etc... De CME de cisco has hecho integraciones con un asterisk?
Se solicita experto en VoiP para dar una videoconferencia a través de skype el día Viernes 18 entre las 9 a 12:00 de la mañana. Solo va a hacer 1 HORA o menos de Videollamada. -Experto en Asterisk y que conozca TRIXBOX y ELASTIX. -Tener conocimientos generales de VOIP. -Conocimientos de Call Manager Cisco o CUCM. -Conocimientos avanzados de LINUX. Opcional: -Conocimientos de Primarios E1, FXO y FXS. -¿Has utilizado un Fonebrigde de Redfone? Está descatalogado pero valoro mucho estos conocimientos
requerido consultor de vicidial, asterisk y mysql para múltiples proyectos. preferiblemente residente en Bogota. debe se experto en el manejo de la plataforma vicidial.
Интеграция с чатом Tidio можно поставить встроенный чат с амо Интеграция формы с сайта (заявки будут приходить в АМО) Будет 3 воронки для (Новый лид, Обработка отказов или ушедших, Активные подписки) обновила файл Подключение и настройка каналов формирования новых лидов: WhatsApp, Viber, Telegram можно без них, Facebook, Instagram Direct, Email, сайт, IP-телефония Ring Central, Facebook рекламные лиды, Instagram комментарии, связка с Acuity Scheduling (системой бронирования) Подключение и настройка почт, для использования внутри АМО (Для переписки с клиентами) Настройка полей и карточек клиента. (нужна будет информация какие данные в полях будут) на данный момент поля такие в acuityscheduling (end time
I am looking for a ready made multi tenant omni channel call centre platform with source code. It should be based on FreeSwitch only. Features of contact centre platform - Dialers - Predictive dialing, progressive dialing, manual dialing Full reports for agents, SIP lines, minutes etc. Complete and dashboards which can be customised on tenant levels as well Full blown feature of Barge, whisper and coaching Different levels of logins - Super admin, admin, tenant admin, supervisor and agents. By omnichannel I mean it should have all integrations of 1. Social media queues - Twitter, Facebook, LinkedIn, Insta and so on 2. WhatsApp API for business 3. SMS integration 4. SMTP integration 5. Ticketing system We would want the complete API stack as well and should be completely bug fre...
Somos una app para realizar llamadas internacionales y envío de mensajería de texto. Estamos en la búsqueda de un desarrollador o asesor que domine Ionic 4 para realizar integración para función de llamadas por internet con protocolo SIP y Freeswitch. La posición es para trabajar remotamente a tiempo parcial o brindar asesoría. Buscamos un profesional de alto nivel cuyo trabajo sea estructurado, en equipo y organizado. El trabajo será remoto, desde su casa, utilizando la herramienta de monitoreo de Hubstaff.
Hello, We have a dashboard that is fed by public data from the social networks of about 600 people. I have a lot of information to collect and I need it for twitter, facebook and instagram. On twitter, for exa...about the impression. 1 - followers 2 - following 3 - tweets last week 4 - retweets last week 5 - like last week 6 - username (we also have it on the base) 7 - profile picture 8 - most used hashtags *9 - latest tweets (img/video + text) *10 - likes on each tweet *11 - retweets on each tweet 12 - creation data of each tweet 13 - bio 14 - updated date/time *15 - comments on each tweet The data with asterisk I need with a lower frequency, if possible every 15 mins. the rest, a few times a day is enough. facebook and instagram are similar, but instagram i need stories/reels an...
Hello, I need some one to help to make any solution to call my client to confirmed the appointment date and time, the system work like this: 1) Upload the data (Mobile Number , Doctor Name , Date , Time). 2) Schedule The Time the system will be start to call the client. 3) the system will use the SIP And VoIP & Any PBX to call client. 4) After call client if the client answer they will be listen massage In Arabic ( Hello , you have an appointment tomorrow with Dr. Mohammed , at 05:00 PM , Please chose 1 to confirm your appointment 2 to reschedule the appointment , 3 to cancel you appointment ) , If now answer will be configure how many time to re call again after 2 or 3 or what ever hour to confirm again. 5) the report must be live , so i can enter the dashboard and see how ma...
Looking for a Asterisk guru that know how to implement STIR/SHAKEN and knows the requirements and process on getting a token. Looking for you to do to all the work including working with my carrier to test before going into production. These just some of the things that need to be done. FCC RMD (Robocall Mitigation Database) Publice Notice RMD link RMD Filing instructions Obtain FRN Obtain OCN
Integrate Asterisk + UniMRCP + AWS LEX
Integrate and test Asterisk+unimrcp+AWS lex
...- * Contact email: email field, validate for valid email. - * Contact phone: phone field, validate for valid phone number with country code. - Operation days: Multiple selection field with the 7 days of the week. - Hours Opening: clock time field. - Hours Closing: click time field. - * Loading / Unloading: Selectable radio group with the values FCFS or Appointment. - Notes: Text area field. * Asterisk notes required field. CRUD Operations to be integrated with REST API: (SEE entities-controller) Grid to present a button for editing the fields of the selected row. On row selected, the details of a given entity should be loaded to the same form used for addition. Acceptance criteria: - github
Tenemos de una página web donde registra las llamadas entrantes de los teléfonos que se reciben y actualmente no funciona. >>>El funcionamiento correcto es: En la web doy de alta teléfono A. Desde teléfono A realizo una llamada al teléfono de Test (éste siempre es el mismo). En la web me muestra el resultado de la llamada (con un "OK") del teléfono A registrando fecha y hora. Y así sucesivamente con todos los teléfonos que de alta. >>>Lo que falla ahora es: Cuando realizo una llamada del teléfono A al teléfono Test, no me registra en la web esta llamada. >>>> Comentario: Esto ya me paso una vez, y el creador de la web me lo corrigió en 10-15 min, pero act...
extend our existing SIP server implementation based on JAVA as an eclipse RCP plugin with full SIP functionality
Hello, We are an independent promotional small company that deal with promotional advertising. I need a bit of help with Asterisk voicemail, configuration and sip. Voices for the Voicemail will be provided, SIP and public business number and whatever else you require. Anyone is welcomed to apply. Due to confidentiality and security reason, everything else will be discussed in chat privately. Thank you.
You may be in any field. I am a investment consultant, we deal in Mutual funds, Life Insurance, health Insurance, retirement plans and motor insurance. I need you to ur friends investment through SIP or take insurance. Instant 10% of total value, transfer to whoever refer me and close the deal. I get only small amount if I sell like this, but it’s fine we grow together!! I will provide end to end training to all interested candidates!! Thanks
Hello! Thank you! We appreciate you taking the time to read this! We’re looking for a pixel artist and...release and wanted to know what the cost would look like to have this looped art created. We’ll also need a background. Setting: A sunny beach. Seagulls, clouds, moving water. All subtle movement in the background is important to us! Character: Yellow skinned male facing away toward the ocean. Animation: picks up a cup of lemonade off of a small table then puts it back after taking a sip ***we’d like the lemonade to move in the glass a little as he puts it back. Turning around to face the camera and smiling. Every now and again we want the face to change into an alien and have a UFO fly across the sky in the background We’d also like the...
Hi there i do e-commerce in GCC countries (Kuwait-Bahrain-Saudi Arabia -Qatar-Oman) and when I use a ZADARMA sip account or any sip account and i use Saudi Arabia number to call a customer from Saudi Arabia my numbers keep getting banned after 60 calls or something around that! do you have any solution because Saudi Arabia blocked (Saudi Arabia's Minister of Communications and Information Technology, Abdullah al-Sawaha, announced on Wednesday that the ban on Voice over Internet Protocol (VOIP) calls will be lifted in the kingdom within a week Sep 13, 2017) let me know if you have any solution for me thank you
We develop Asterisk / Freepbx graphical ui and add more our own functions.
...call is answered IVR sound file is heard and the customer is asked to press 1 to speak to an agent or 2 to end the call and allow them to give us a call back when they have time, if we call multiple people and they all press 1..they will need to be put into a queue if all of our agents are busy. Must be able to easily load new audio files to use in the IVR. Needs to be compatible with FreePBX/Asterisk, ideally an easy to add module but I am flexible. Scenario After customer presses 1 they are connected to the agents extension, when agent requests the customer to input our verification code, it'll show up on the agents webGUI/Dashboard. I'm happy to listen if you have a better idea to make this more user friendly. - must have the ability to receive the input thro...
...hectic fast moving routine a casual friendly neighborhood cafe. Amidst many garage themed cafes here comes the original cafe that has been set up in a garage breaking all the stereotypes. Refreshing outdoor environment with mountain vibes where you can have flavouful foods and beverages while balancing the Jenga blocks. Fun filled quality time can be spent with your friends playing foosball with a sip of Mojito, on the flip side as your health freak buddies enjoying our Steamed Hot Kozhukattais. Surprise your loved ones with our grandeur decors making them feel so special on their day by throwing a private party." We need menu description and write-ups for our cafe...
Hi there i do e-commerce in GCC countries (Kuwait-Bahrain-Saudi Arabia -Qatar-Oman) and when I use a ZADARMA sip account or any sip account and i use Saudi Arabia number to call a customer from Saudi Arabia my numbers keep getting banned after 60 calls or something around that! do you have any solution because Saudi Arabia blocked (Saudi Arabia's Minister of Communications and Information Technology, Abdullah al-Sawaha, announced on Wednesday that the ban on Voice over Internet Protocol (VOIP) calls will be lifted in the kingdom within a week Sep 13, 2017) let me know if you have any solution for me thank you have a nice day
Our existing Asterisk node will be migrated into a K8s setup plus load balancer setup (Kamailio). We would like to outsource this project!
