Sip spoof call工作

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    9,402 sip spoof call 搜到的工作,价格货币为 HKD

    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    voip语音网关模块 已经结束 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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    Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.

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    VoIP Video IM等即时通讯功能。 支持Android,IPhone,Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。

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    ...8、需要提供所有源代码和演示版本(演示版本的运行平台包括android、IOS),以及可交付的中转服务器; 9、并不需要完整的语音通讯软件,只需要能够实现语音传送、播放的功能模块即可。 10、演示内容:A用户点击“通话”按钮,中转服务器可通过设置好的IP地址,找到B用户,并发送通话请求。B用户接受请求后,双方可以进行通话。 11、你可以采用第三方组件,例如SIP,但是需要告诉我们是哪家的,效果如何。因为考虑到我国网络问题,所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到,叫IEMAKER,通过安卓盒子,可以实现直播,但是目前还不能实现声音直播,因此,需要开发这个模块,用于该软件。

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    We need a powerful photo liveness detection. User will send a photo and we should detect if photo is spoof, fake or live and real. Perhaps users will send a photo from their mobile screen or a printed photo on paper to fraud. We should prevent user and this is vital for our business. I hope that a good artificial intelligence or machine learning programmer

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    Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.

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    Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    AXNET VoIP 3 天 left

    I am Peter from Rwanda/ Africa. I am looking someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones

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    Hello, We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough

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    Trophy icon Study Cafe Name is needed!! 5 天 left

    NAME IS NEEDED. NOT A LOGO!!! I NEED A NAME SUGGESTION Study Cafe Name needed! - This is a space where you can study and get sip of coffee and some desserts. - space is mainly for studying (and maybe working) - there are space for them to do team work - there are space for customer to rest too - please see some example pictures.

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    48 项参赛作品

    I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.

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    We are looking for an artist to create desirable pieces that the general public would like to sign up and...instructions to replicate which we can share with our local artists to replicate in a classroom setting (photo+text). Hoping for quality work very similar to the top paint and sip companies out there, and would be willing to work with long term.

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    1 项参赛作品

    ...an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5. - The SIP signalling shall transport a 180 RINGING message without SDP parameters meaning the originator should

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    ...platform initiates an outbound call then sends an SMS if they customer does not answer. Here is how the flow works 1. A lead is posted to the API 2. Call is initiated via Twilio Outbound Voice API using a Phone Number. 3. When call is answered, the call is forwarded to another Twilio phone number that uses Elastic SIP Trunk. It is attached to our IBM

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    We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    kamailio5 SIP confing with RTPproxy 结束于 left
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    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [登录来查看链接] if you can make it please bid

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    Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

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    I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

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    i need to send call server to local voip gateway like server in USA VoIP gateway in Bangladesh server to Bangladesh Openwrt Router connected BY VPN and VoIP gateway connected to openwrt router so i send call from server its work fine but i need some one who can help me to create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but

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    I have a bunch of phones ...firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they come up and provision.

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    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

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    WMI Spoofer 已经结束 left

    Create a desktop application which can spoof wmi information

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    ...[登录来查看链接] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order

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    Application key features: * high definition audio and video calls * audio c...sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers allowing to reach everyone that has a "classic" phone line.

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    Homer Seven SIP Capture 已经结束 left

    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

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    I need an illustrator 已经结束 left

    It is a short adult humor childrens book spoof. Looking for somewhat rudimentary drawings, pencil drawn stick figure style images to enhance the humor and create a truthful honest look at homelife with kids.

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    FreePBX Setup 已经结束 left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

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    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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    i need a software to remove or change SIP RTP PayLoad like if i install this software on os it have to remove or change all incoming and outgoing SIP RTP PayLoad if you can make it please let me know

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    Mobile VoIP Dialer 已经结束 left

    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    solución pbx 已经结束 left

    Tengo una centralita pbx con issabel y 2 troncales sip con gateway goip en redes difefentes pero siempre se desconectan y tengo que reiniciar el servidos pbx para que se vuelvan a restablecer

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    Design project 已经结束 left

    Sip and paint artist needed for one week event

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    need hobby geeks 已经结束 left

    Looking for an On the Call Technician, who has some experience in customer service, We will provide a Voip Phone Number to handle inbound and outbound calls, Earnings will Vary pending on skills and experience, and may be ongoing. Jobs Requirements, [登录来查看链接] Have Knowledge in all Operating System Platforms. [登录来查看链接] to make a Sale 3. Willing to answer

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    Hi Dear In running local FreePbx server under elastix and the server have 2 network card One card link to Sip line from local Telecome company and another network card for internet connection My local server is online now via port forward from the router and it’s working by Zoiper app very fine in computer an mobile... in local network and online

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    Hi Dear In running local FreePbx server under elastix and the server link to Sip line from local Telecome company and there is another network card for internet connection My local server is online now via port forward from the router and it’s working with Zoiper app very fine in computer an mobile... But the problem its just working with Zoiper nothing

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    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or phonegap or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through a SIP proxy. we will provide all the server details needed

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    ...agent and the agent wants to end the conversation by pressing the key 0 in the conversation the connection is terminated and the agent hears a end pROMT many thanks for their call. The software should have the following possible operator channel control per ivr to announce add phone number or external phone number for mediating with listening directly

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    i need to configure 3cx and add sip trunks

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    I want to add new SIP domain in skype for business and make it working.

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    VoIP SIP Intercom Module 已经结束 left

    ...Management (Multi-Partition File System, Advanced Firmware Management, Bootloader, Secure Area) SIP Fusion TCP/IPv4/IPv6 DHCP Client Extensible Command Line Interface Web Server with Extensible Configuration Web App and Configuration and Control Web Services VoIP Call Manager (includes support for incoming calls, outgoing calls, hold, conferencing, transfers

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    Art History Essay 已经结束 left

    ...Today our fans want to ask you a few questions about some of your paintings and the media you have used. Daumier: Well of course. Daumier picks up his water and takes a small sip. Presenter: The first painting our viewers are interested in, is your print Gargantua. Daumier clears his throat. Daumier: Ah yes, one of my finest caricatures. Presenter:

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    ...general familiarity with backend frameworks likes Rails, Django, Phalcon, YII, etc. Fluency in SQL Databases, NoSQL Databases, Cloud Storage APIs, etc. Fluency in TCP, XML, SIP and other web protocols & patterns. Experience with building secure web applications, troubleshooting, well scalable applications, etc. Extensive experience with API architecture

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    Dears Developers We are ruing Sip service under FreePBX platform And we use [登录来查看链接] service to keep our server online using router port forward . So we need app with SIP protocol support to make viop calls with our Labled & Customization with Full Branding Mobile Dialer support IOS iPhone and android mobile. Mobile Dialer FEATURES 1-User friendly

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    ...hire new people for the NOC, so ongoing training is important to us. Here are some suggestions as to what the training should cover. We want to focus on Voip and SIP. Why Voip SIP - protocol structure, format, Commands, how to read it, Error codes, interpreting the error codes. Under Voip introduction to Voip Why use Voip over Analog How

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    the scenario; - I have sip phones registered to my ip pbx - when a sip phone try to call a mobile/landline - my pbx will route the call to "skype gateway" - skype gateway will use a skype account to connect the call transparently sip phone -> ippbx -> skypegateway -> gsm/landline inbound calls to be handled with ip pbx the project sho...

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