Voice over Internet Protocol or VoIP refers to a host of transmission technologies meant to deliver voice communications over IP networks that include packet switched networks and Internet. VoIP systems include communication services such as facsimile, voice, and voice messaging applications. A number of firms are shifting to this system to promote a cost effective medium of communication. You can hire an expert who can help your firm to leverage the potential of this new system. Just post a job today! VoIP Developers 분야 채용

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    39 건 검색 완료, 금액 기준 - USD

    I am looking for SfB experience. We have : Instant Messaging/Chat Desktop/PPT Sharing File sharing Audioconferencing Dial-in Audioconferencing(Masergy/Toll free) Voice Mail With Exchange SBC - SIP trunk/PSTN Exchange 2013 integrated for Voice mail, Resource booking Open External Federation Contact me, if you have experience back end troubleshooting experience.

    min $50 / hr
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    Hello All, We are looking for CCNP or CCIE Voice engineer with a wide experience in CUCM 11.5 configuration and implementation of: - CUCM - Installed but not fully configured - CUC - Installed but not fully configured - CIM&P - Installed but not fully configured - Expressways E and C - Need installation and configuration - Voice Gateways (ISR 3925) - Configured Following Cisco Best pract...

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    3CX autodialer 6 일 left
    확인

    Hello, we need to create autodialer on 3CX, which will take number from another system via API, call to number, play message, receive client pushed button(if you happy for our services press1, if you not happy press2) and insert value back to our CRM system via API. in front your bid please wrote 3cx, i would like to know if you even read my project post. thanks.

    $240 (Avg Bid)
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    Hello need to achieve push notification by installing a voip script in my server and add my voip pem file to the script so it wakes up the app script it here and need to use curl command [로그인하시면, URL을 확인하실 수 있습니다.] I use fusionpbx and i will provide voip pem file wiht passphrase only bid if you are technical competent to execute this and have doen it before.

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    Using Twilio API VOIP 6 일 left
    확인

    Hi We are using Twilio API for an app. The problem is that after 10:53 of communication both speakers can't here each other anymore even though the call is still going on. The app works on iOS. We are astonished by the fact that in all our tries the call automatically cease to function after this exact time. We are looking for someone with an expertise on the issue. Please find attached...

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    freepbx configuration 6 일 left
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    I have installed new freepbx version from site: [로그인하시면, URL을 확인하실 수 있습니다.] I don't succeed to connect an extention I also have old pbx that we can backup configuration there and copy them to new pbx

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    Voip Opensips 6 일 left

    i need [로그인하시면, URL을 확인하실 수 있습니다.] With modules callcenter dialog CDR load balance NAT will Work With opensips-cp

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    Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

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    I'm looking for somebody to help me setup a fresh new installation of FUSIONPBX with special attention to Branding setup, incoming/outgoing call routes, phones provisioning and features like IVR, Fax, paging, etc. Preferable if speak Spanish. The help/works has to be done thru Teamviewer platform. I calculate 2-4 hours first week and one hour next 3 weeks. Total approx. 10h.

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    je recherche à acheter, a louer ou à faire créer un predictive dialer pour les callcenter

    $139 (Avg Bid)
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    We are looking for a virtual phone solution to be accessed with iPhone that can have an auto attendant (option 1: Sales and 2: Support). I need to build our own solution, especially we don't receive many calls. I expect that the solution would be in the cloud. The system will have a US number. The developer may utilize technologies and system components like soft PBX, SIP, Asterisk, DID, etc....

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    I have flexisip , Asterisk , Opensips working well . I want experienced people flexisip , Asterisk , Opensips Who can support configuration document. Client <--->Flexisip ( Frontend )<---> Asterisk ( Backend )<---->PSTN Please check this link and If You can configure and support contack to me. [로그인하시면, URL을 확인하실 수 있습니다.] [로그인하시면, URL을 확인하실 수 있습니다.] [로그인하시면, URL을 확인하실 수 있습니다.]...

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    my need voip billing switch with unlimited customer agent level,unlimited rate plan trunk, automatic call quality control (asr acd) like sippy switch,automatic buyer & seller account create & ip add option,latest hacking security protection for user access & ip allow, anti block dealer with whats apps facility for all country & main is Saudi Dubai .my need ready product so if you ...

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    Use Agora voice API to create an android application through which we can communicate.

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    Hi there, im looking to get some assistance with configuring a Cisco router and some WIFI Access points (4). Needs to have some security as well as having a guest wifi with limited/timed access.

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    Hello, We are VOIP service providers. Currently looking for a software which can generator voip calls and terminate them to any country with variable call durations The catch here is that if the calls does not come back to the originator platform without a specified time frame it should automatically get dropped/disconnected. Along with it IVR which are customized and audio files which can select...

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    Need assistance with the following 1) Configure Dialplan on ASTPP Fswitch for inbound and outbound call routing 2) Setup all aspects of ASTPP and Fswtich. 3) Must be able to work using remote share enviroment.

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    app caller id changer 3 일 left
    확인

    app caller id changer based on SIP, for example [로그인하시면, URL을 확인하실 수 있습니다.] [로그인하시면, URL을 확인하실 수 있습니다.]

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    We need 3cx PBX remove limitations with multi-tenant and use following modules use, Call Queues, Call Reports, Call Recording, Contact Center, Call Recording Control, Inbuilt Failover, Standby License basically we need crack enterprise. anyone have or done already ?

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    We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outb...

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    Asterisk sip consultant 2 일 left
    확인

    Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work

    $16 / hr (Avg Bid)
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    Looking for an expert to integrate Vtiger with Asterisk FreePBX What Exactly I need is: 1.Click-to-call right from a lead or contact record to save time. 2. See the callers' contact information on the screen whenever they call 3. Quickly create opportunities and contacts right from the incoming call popup 4. All calls are logged so that you can refer back to call histories later 5. Automatic...

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    Hi Team, Our centre has the best-in-class environment with state of the art infrastructure, currently there are 20 well trained associates eagerly waiting to deliver their expertise. We provide assistance with Lead Generation, Inbound/Outbound Sales, Lifestyle Surveys, Appointment Fixing, Customer Support, Technical Support & Email/Chat Support. We have the space and capacity to expand the nu...

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    i am looking for a system that can block all voip and VPN protocol like pptp,l2tp,sstp,openvpn etc also we want to block all voice apps like whatsapp,imo,viber etc we install this on our ISP if you need any router we can manage that we have right now more then 800G BW load so project only for expert not for tester we want to make something like china firewall

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    Lua Prosody Xmpp 1 일 left
    확인

    Lua scripts and xmpp prosody

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    1С:Предприятие 8.3 (8.3.13.1513) Альфа - Авто: Автосалон+Автосервис+Автозапчасти ПРОФ, редакция 5.1 (5.1.13.04) Локальная версия. 3 пользователя подключённые по сети к 1 компьютеру. При поступлении звонка от номера который есть в базе Звонок должен быть автоматически зарегистрирован на "Ресепшн" указан клиент который есть в базе Сотрудник (сотрудник или из базы телефона кто брал трубку и...

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    when try to log web VSC 3 its get error An error has occured. If it will repeat again please contact the administrator. Error reference number: f803707165ea4ecaa624e65adf4ec72b.

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    Hello, I need some help in Asterisk IAX2 trunking and callflows over it. Experience in the fork Elastix or Issabel is a great plus. Will be a half day job or a bit longer. Kind regards, Danny

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    I want to develop a application that can do real time call transcribe in a call center environment, you must provide your description of solution in detail on how you can push the audio stream from a traditional call center architecture (IVR/ACD) or in open source environment like Asterisk. how to classify which audio stream belongs to which agent. Currently we already developed a demo code whi...

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    We have freepbx running in production but would like to have the server optimized. Here are few things for a start that need to be done: 1) Securing freepbx server - fail2ban, iptables etc ipv4 and ipv6 2) ensuing that secure channel is used - iax, tls etc 3) ensure that the SIP setup is optimal 4) ensure that codecs and sip settings are right 5) configuring DID for extensions 6) Ensuring that bac...

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    Hello, Need to integrate natural language speech recognition (NLSR) technology or twilio IVR , Speech to Text API for Taxicab Dispatch system , sending speech to my computerize dispatch system and creating pick up trips for drivers . Thanks

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    WebRTC, VoIP, PJSIP, Opus 14 시간 left
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    I need you to develop calls functionality using VoIP, PJSIP, WebRTC, and Opus. Please bid only if you have prior knowledge in these fields. Type "call" in proposal so I know you have read project description.

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    Hello i need open wrt router firmware solution for voip gateway to call passing cloud server via open wrt router like tplink 3420 model & other open wrt router firmware support model with udp tunnel udp header with rtp [로그인하시면, URL을 확인하실 수 있습니다.] you have complete this project or you thinking complete possible developed for me contact please.

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    I am looking for a technical assistant with knowledge and some experience in cisco, voip, network . Reliability, discipline is more important besides technical experience. I need someone locally in Mumbai.

    $367 (Avg Bid)
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    Tele Calling 8 시간 left

    Hi, We are looking for short term project. This is calling project.

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    Hey, We are looking for a Asterisk / Vicidial Expert who have good experience and is knowledgable and familier with all the ins and outs of vicidial to setup a cluster. Also you should be able to start ASAP. Thanks

    $213 (Avg Bid)
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    Hey, We are looking for a Asterisk / Vicidial Expert who have good experience and is knowledgable and familier with all the ins and outs of vicidial to setup a cluster. Also you should be able to start ASAP. Thanks

    $5 / hr (Avg Bid)
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    Hey, We are looking for a Asterisk / Vicidial Expert who have good experience and is knowledgable and familier with all the ins and outs of vicidial to setup a cluster. Also you should be able to start ASAP. Thanks

    $6 / hr (Avg Bid)
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    I am looking to configure openVPN on a yealink T41P. I configured on a yealink T46G but I am having issues with the T41P. I am using PFsense as the openVPN and it is configure already. I just need to learn how to use it on the Yelaink T41P. Please let me know if you have done this VPN with the T41P. Thanks

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    VoIP 커뮤니티 게시글 (최상위)