Viber sip asteriskεργασίες
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
an app for CRYPTO SIP INVESTMENT
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!
I have the same requirement with https://www.freelancer.com/projects/python/python-sip-voip-client-with/?ngsw-bypass=&w=f. 1. I need a SIP sample application written in python3 gstreamer, running on Linux OS. 2. Call peer, and its sink/src are flexible. (nice to be appsrc/appsink since I already have a gstreamer app need to communicate with) Thanks!
Hi Md. Rakibul Islam R., I noticed your profile and would like to offer you my project. We can discuss any details over chat. Are you good at customizing freepbx/asterisk server? Any experience creating an IVR system that can run a call campaign etc Do you have telegram as it is easier to talk there
I need someone who have good knowledge of help to set VOIP apps and I already have sip account at flowroute . We usually do manual call need to set up my custom number with my voip and explain work flow .
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
WE are looking for a technician to execute SIP Trunk configuration for CISCO router and integrating with existing Panasonic KXTDE pbx. Cisco router model is Cisco ISR 4321 Bundle, w/UC License (ISR4321-V/K9)
Hello... we are looking for opensips developer for modifications in sip header. We want to comply our opensips based switch with stir/shaken standards and need some expert who can make modifications in our present software which is based on opensips 2.2 version.
I NEED AN ASTERISK PBK DEVELOPER for my project..We need an expert with Asterisk/FreePBX in building a phone system for my project. Front end is developed and need to integrate the ASTERISK from the scratch. Must have a strong knowledge of ASTERISK & already integrated the system earlier.
I am looking for someone who can show me how to get the res_speech_gdfe module in Asterisk working... I have spent so many hours on trying to get this to work, I think it is smarter to find someone who has done this before and is willing to to get paid to show me on how to do this correctly and show me the error of my ways. Rgds, Gertjan
YOUR REVIEWS ARE GOOD THE ASTERISK DIALER BOT YOU SHOW HAS ALL FEATURES WE NEED. SOME FEATURES WILL MAKE SOME CHANGE. WE WANT TO SPEAK WITH YOU AND MAKE GOOD BUSINESS
I am looking for someone who is expert in c, sip protocol and networking skills to help me with my job. This will be an ongoing project and that person will need to spend around 4 hours a day preferably during US east coast time zone. I am expecting that person to help me with my job. I need someone who has good experience in c and pj sip library This is to help me complete my job. I am looking for someone who will complete the tasks assigned to me in my job. I will be paying you on per hour basis and it will be 3 to 4 hours of work per day. If more time will pay for that. But generally it will be around that and that person needs to be logging into my laptop remotely using zoom or any desk kind of softwares and work as the code can be accessed only with vpn access are you c...
Astrisk based Freepbx virtual appliance setup on my ESXI and setup sip trunk & 30 extentions. my sangoma box is faulty so need freepbx expert to install setup freepbx as VM and do the config.
Hola, me gustaría poder realizar transcripciones en tiempo real con el servicio de Speech to Text de IBM y Asterisk ¿Me ayudan?
I want to be able to resell sip trunks at a reliable price, anyone with knowledge in this industry might be able to suggest whole sale companies?
Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Kamailio such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.
...and grow together! We are looking for an agile developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different techologies by self learning. We are looking for attention to detail, communication and reliability. Mandatory expertise Nodejs Angular React NestJs Websocket Mongo OPTIONAL Sails Ionic Openvidu/mediasoup Asterisk Chatbot php symfony !NOT NEGOCIABLE! You are expected to work in GMT+2 timezone. We pay 2'500CHF per month (can be reviewed depending on seniority) and offer 20 days of paid holidays per year. TO APPLY 1. Let us know your experience in years and short description of your work performed on all mandatory elements and ideally on optional elements. 2. Send your CV 3. Put your ...
Configure asterisk/freepbx with dongles attached to receive incoming sip traffic and dial out through attached dongles.
I run an ISP & WISP using Mikrotik routers and Ubiquity WiFi equipment. The backbone of the network requires a rethink to allow for better failover, maximise performance, improve routing processes and load balancing. I had some issues with SIP services however the cause of the issue was found and resolved in the short term. I am also considering migrating client connections to IPV6 as they all sit in private address space. I would like to engage a Mikrotik engineer to assist the process. There is 1 X Mikrotik connecting to 3 X Gbps ISP fiber. A Ubiquity Air Fiber link to a Mikrotik at a central distribution point feeding to the Ubiquity WiFi network and further Mikrotik connected to 2 X DSL. 1 X DSL connection is a fixed IP hosting web services.
I need to install an Asterisk server and play a mp3 file to a list of numbers.
please please read the attached description till i...study-abroad destinations, and immigration destinations, and study-for-less destinations) -scholarships topics page -webinars -library (selected blogs and videos) Footer: *our offices: addresses with telephone numbers and locations and working hours, *social media (FB,twitter,linkedin,google,youtube,instegram), *our contact info (telephone, mail, WhatsApp, viber, telegram, diferent inquiry/contact forms) Central services contain 3 services of a total of 8 services on the website: A*World immigration and cross-border service (multi-profile) B*Book your exam service C*World travel preparations and arrangement (the shop) For those who are interested, please read the attachment till ...
I have Vtiger 7 CRM, previously we had to use asterisk PBX installation for the specific setup but due to the inactive time period of the business, existing PBX settings are no longer exist. we did install issabel Asterix into another centos machine. since we already have all installations for both issabel Asterix and vtiger, and they are both in working conditions and vtiger already has the related module for its own... it should to be taken only 1 hour for a guy who has previous experience on the task... (both voip calls and vtiger are in working condition) What I want: -CRM Users should able to use "click to call" dialing options for making outgoing calls right from Vtiger CRM by clicking on the numbers. -Instant pop-up alerts for incoming calls displaying the cont...
Hi Saradha R., I noticed your profile and would like to offer you my project. We can discuss any details over chat. my viber or whatsapp no +358451242062
I'm looking for someone that is willing to answer questions that I have about Asterisk / FreePBX configuration, troubleshooting, etc. Should be well-versed in Asterisk, Linux, SSH, SIP, etc. For example, voicemail email notifications are being sent from asterisk[at] and my spam filter does not like that. I tried editing the config via FreePBX >> Voicemail Admin, but it's till not working. So I need help getting that fixed and maybe setting up the proper DNS records for it to use one of my FQDN's. Note: My distro uses Sendmail and I think Postfix would be better. Note: If we decide to proceed with this job via your Freelancer.com bid, payments will only be made via Freelancer.com. Kindly do not make an offer if you plan to ask to b...
I need help to amend current Perl script to work with new version of Asterisk PBX system. I need someone who has knowledge of asterisk PBX system.
Need multiple stick plans converted/modified to SIP. Looking for a long term working relationship. Will initially provide one set to be converted to evaluate process and delivery before requesting additional plans to be converted.
I have just set up Freepbx for my home business.. I have the extensions set up and working, but need assistance with the inbound and outbound settings I am using mynetfone as the provider of the 2 SIP lines I need to connect & configure them to the system.
Hi, I have freepbx installed in cloud vps. There is an issue with outgoing calls.
...I want these labels to be bright, colorful, and eye-catching to the customer. Below I attached what needs to be on the label along with images and styles I like. THE LABEL SIZE IS 8x5 inches. LOGO IS ATTACHED. Strawberry Guava No Caffeine Net Weight: 3 OZ As tasty and satisfying as pure fruit juice; with a tropical paradise twist that is absolutely bursting with fresh island flavor! At first sip your taste buds will wake up to a bold fresh guava flavor that fades to a sweet pineapple and strawberry flavor! Ingredients: Candied pineapple pieces (pineapple, sugar), candied papaya pieces (papaya, sugar), hibiscus petals, elderberries, beetroot pieces, natural flavor, strawberry pieces, guava pieces, vanilla pieces How To Brew: 2 TSP Per 8 OZ 212 F Steep for 8-10 Minutes CONTAI...
...had to use asterisk PBX installation for the specific setup but due to the inactive time period of the business, existing PBX settings are no longer exist. we did install issabel Asterix into another centos machine. since we already have all installations for both issabel Asterix and vtiger, and they are both in working conditions and vtiger already has the related module for its own... it should to be taken only 1 hour for a guy who has previous experience on the task... (both voip calls and vtiger are in working condition) What I want: -A single click to call customers: Initiate calls right from Vtiger CRM by clicking on -Instant pop-up alerts for incoming calls displaying a contact name -CRM Users can use single-click dialing options for making outgoing calls. -Vtiger Ast...
We are looking customised multi-tenant open source call centre VOIP solution, with an android and IoS app for calling agents. We should be able to appoint resellers, white label it for resellers, we should be able to sell b2b c...Dashboard DID management Graphical reports Multi-level IVRS Productivity Sale Graph Real-time call status Standard Call features Standard reports and other range of reports Voicemail All basic features Inbound number Black Listing Email & SMS Module Internal Chat Module Skill based Routing with Agent Ranking Sticky Agent You may use any or many open source softwares to get these features, like Asterisk Fusion Pbx, ASTPP, FreePbx, free switch telephony, webrtc, a2billing, vicidial, magnus billing etc. The system has to be robust scalable and capable to...
