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    19,587 asterisk custom voip configuration робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

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    Texte zu Hunderassen in deutscher Sprache verfassen 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....

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    In our VPS we have problems with the delivery of mail: Some arrive at the SPAM tray, others do not arrive and others simply bounce. And these bring us many complaints from our customers. Hello, In our server we have problems with the delivery of mail: Some arrive in the SPAM tray, others do not arrive and others simply bounce. I often receive calls telling me that: - The emails they send do not...

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    JSP JAVA expart Need FOR VOS3000 VOIP LINUX Server 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $55 / hr (Avg Bid)
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    Customization of LinPhone SIP / VoIP Client 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    VOIP yate call pass 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    VOIP phone system 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    we need a phone system based on twilio platform.

    $680 (Avg Bid)
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    AWS WAF and AWS Shield configuration 6 дні(-в) left
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    Need to configure AWS WAF and Shield configuration on platform running with ELB

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    Bitrix integration 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    integrate bitrix with asterisk system -

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    Develop a new web service for managing virtual notes. 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...marked with an asterisk (*). Backend API (api/) api ├── Dockerfile # Description of Docker image ├── [увійдіть, щоб побачити URL] # NPM dependencies ├── spec # The backend tests ├── src │ ├── [увійдіть, щоб побачити URL] # Express application definition │ ├── config │ │ ├── [увійдіть, щоб побачити URL] # Database connection co...

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    Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

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    nginx configuration 5 дні(-в) left

    No matter what the visitor types [увійдіть, щоб побачити URL] or [увійдіть, щоб побачити URL] or [увійдіть, щоб побачити URL] i want all of the requests to be redirected to [увійдіть, щоб побачити URL] currently i have managed to redirected requests that are coming from [увійдіть, щоб побачити URL] and [увійдіть, щоб побачити URL] to [увійдіть, щоб побачити URL] but i can't find a solution fo...

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    VOIP Project 5 дні(-в) left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    FreePBX support 5 дні(-в) left

    Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from

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    asterisk expert needed 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

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    SIP To SIP register over Masquerade VPN 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [увійдіть, щоб побачити URL] or yate or besip [увійдіть, щоб побачити URL]

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    VOIP or Live stream for my website users 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    Step 1: Homepage with logins Step2: Login page with details step3: VOIP or Live Stream features

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    Integrazione api voip 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

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    Paid VOIP services app 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

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    configuracion de internos y de central asterisk

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    Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem

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    ...documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording •...

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    Template and Invoice Maker 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    To permit clients to fill out forms and documents on our...for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already [увійдіть, щоб побачити URL] is a Linux only project !

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    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

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    Opensource VoIP needed to replace SINCH on iOS App 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    **Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development

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    FreePBX, WebRTC (UCP Web Phone) 1 день left
    ПІДТВЕРДЖЕНО

    I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?

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    speed optimization and CDN configuration 1 день left
    ПІДТВЕРДЖЕНО

    Hi dears My current loading time for my website is 13 second I need to be less than 3 second without ny effecting on the theme and current plugin. I'm using Woocommerce and this is my store [увійдіть, щоб побачити URL] Thank you Saeed

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    ...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message

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    Hi, i would like to cooperate with an expert on FreeSwitch configuration. Just to make it straight, I don't plan to use any web interface as a plugin extension like FusionPBX etc. I want configurations from the filesystem. Currently, I just have two questions. But the person that will help me on those items i plan to hire with a monthly consultation

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    123 flash chat configuration and mobil chat app setup. 1 день left
    ПІДТВЕРДЖЕНО

    hello, im looking some one talented, who can setup 123flash chat and make android and iphone application for mobil chat. only experiment guy's please!

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    Voip Calling App -- 2 1 день left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    Asterisk & Zoho CRM 1 день left
    ПІДТВЕРДЖЕНО

    We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information

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    ...availability 24-Marketing automation 25-Goal-based nurturing 26-Salesforce integration 27-Smart content 28-Attribution reporting 29-A/B testing 30-Custom revenue reporting 31-Custom event reporting 32-Custom event triggers 33-Predictive lead scoring 34-Contacts reporting 35-Company reporting 36-Event-based segmentation 37-FSBOS Craigslist / Real State

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    voip语音网关模块 17 годин(-и) left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Issabel Features 15 годин(-и) left

    We Need somene that have some skills over ISSABEL Platform VOIP , Asterisk and Call Center Platform . We need to implemented some news features into reports database .

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    Voip Calling App 13 годин(-и) left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    asterisk ari recording 12 годин(-и) left
    ПІДТВЕРДЖЕНО

    I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...

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    freepbx asteriks voip phone system server -- 2 12 годин(-и) left
    ПІДТВЕРДЖЕНО

    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    CONFIGURATION/HOSTING 7 годин(-и) left

    CONFIGURATION/HOSTING............................

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    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Soft Switch development. 4 годин(-и) left

    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

    $9 / hr (Avg Bid)
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    Soft Switch development 3 годин(-и) left

    Asterisk based Soft Switch [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Ajuste e atualização do site 3 годин(-и) left

    Atualização do site, segue as atividades| 0.1. Conversor de tabelas para Rocket (central asterisk) | 0.2 Integração entre o Rocket (central asterisk) e PROSPECTS (area administrativa) | 0.3 Inclusao do Te Ligamos (click to call) e botao blog no menu area cliente do site 1. Blog (do 0) | 2. Integração com Rd Station | [увійдіть, щоб побачити U...

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    La RCSOft software specializzata da anni in sviluppo software gestionali, per potere integrare al suo interno un CRM richiede una interfaccia con il Centralino Voip 3CX v15 in poi, per potere effettuare delle telefonate in uscita e per potere recuperare il numero chiamante in entrata in modo da posizionarsi sul form del cliente. L'interfaccia deve

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    webrtc asterisk Закінчився left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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    ...gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with other freeswitch, asterisk and fusionPBX installations. The task is to look at my setup, determine the problem

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    help with haproxy configuration Закінчився left

    We have haproxy running as our load balancer, and sticking sessions by port. Unfortunately, the queues are too big and users are unable to access backends, need to adjust configurations somehow.

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    Dinstar GSM gateway VOIP Закінчився left

    I have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

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