Android sip gsm工作

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    14,195 android sip gsm 找到的工作,价格在 USD

    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

    $1180 (Avg Bid)
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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.

    $1532 (Avg Bid)
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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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    1、开发一个用于移动APP软件的即时语音通讯模块,发送端是IOS或者android平台,接收端是android平台; 2、即时通讯模块功能主要实现单对单语音即时聊天,无须实现视频聊天; 3、要求操作简单,点击“通话”功能按钮即可实现通讯; 4、由于即时通讯双方可能位置很远,因此必须经过互联网传输,需要中转服务器进行语音媒体的中转。中转服务器可以采用成熟的第三方软件或者协议,但是在项目完成后可以交付给我们; 5、发送方网络的最低可能是ADSL 2M。但是延时不能超过1S; 6、语音质量要高,不能有杂音、回音等问题,需要可以用于商业行为的那种; 7、需要有可供调用的组件和调用方法,我公司可通过调用方法进行上层功能的开发; 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...

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Industrial design 6 日 left
    已验证

    To design the smallest wristband that can accommodate: Wi-Fi, NFC, GSM modules and battery for 24h use.

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    Freepbx and Asterisk settings 6 日 left
    已验证

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    Regain Mailer 6 日 left

    Attached is a word doc that i need formatted into a mailing document that i can send through the post. There are two uses on t...the cut paper - both front and back. Whether it is done in word or other format doesn't matter as long as i can print it off my office Xerox printer which handles up to 300 gsm. I need to open it in my ms office etc program

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    I want to do a real-time iot app. The platform will be parse server. 1-) The sim7000e (simcom) gsm module connects to the arduino internet (to be programmed with nodejs). 2-) Arduino Parse will send data to server platform. (to be programmed with nodejs) 3-) Client will monitor the data in real time. (to be programmed with nodejs) 4-) The parse will

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    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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    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

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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.

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    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 4 日 left
    已验证

    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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    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    i need code that switches over tracking services...if gps shield does not have service use gsm sim longitude and latitude to update tracking...to MySQL database

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    Actionscript 3.0 SIP 4 日 left
    已验证

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    PHP SIP client 4 日 left
    已验证

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project 4 日 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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    Must be proficient in EE and software development. I am looking for someone to build a concept within the IOT space. This will be utilized to monitor d...someone to build a concept within the IOT space. This will be utilized to monitor different devices within the healthcare industry. I am look to have an adhesive translucent GSM based module built

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    ...support for NGOs and association‘s meetings. Help for student Masters or thesis or study of a project, for students, researchers and professionals. arfaouisouhail22@[login to view URL] - gsm: 22 046 269...

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    本地
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    Create SNOM Phone Dial Plan 3 日 left
    已验证

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    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 [login to view URL] or yate or besip [login to view URL]

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    add sip trunk to elastix i have sip trunk from STC

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    Looking for a Linux sysadmin who have experience with serial port settings, serial communication. I want to use Neoway M590E GSM/GPRS modem with USB-UART serial converter, looks like everything okay, but I can't communicate with the modem over terminal. I already tested the communication with the modem under Windows, everything fine, working. You need

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete 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 &bul...

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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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    It is a module that have to comand a relay to switch on/off 220Vac (80A) by GSM.-

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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    Mobile GSM servis, specialized in apple products

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    I need to read the serial communications from a GPS NEO 6M card on a Arduino GSM 1400. GSM 1400 cannot use SoftwareSerial so you would need another method. Project will be an .ino Arduino file.

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    Rebranding of Linphone 已经结束 left

    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    Ringless Voicemail 已经结束 left

    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www

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    I want someone to wrote me a detailed Paper explaining LORAWAN, SIGFOX and NBIOT and the role they play in current IOT market. Differences between GSM and GPS trackers also must be included in this.

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    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

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    Preciso de algurm que trabalhe com programação arduíno. Para programar sensores que captam o som, e o módulo gsm

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    the project is to build ionic app plugin out of linphone sip framework or linphone-cordova-plugin. Plugin should have following features. Audio calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Call History Echo Cancellation Call quality indicator Secure user authentication : md5 / SHA256

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    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

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    Design project 已经结束 left

    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

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    webrtc asterisk 已经结束 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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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the 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

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    Dinstar GSM gateway VOIP 已经结束 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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    Looking for an arduino expert,who can design a wearable wrist smart band with multiple features , features are 2.5 inches screen , gsm anteena, temperature sensor, multiple microcontroller and sensors will work simultaneously. then create API of the product that will be interact with smartphone application and web services Also required design

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    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

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    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    FreeRTOS on LPC2148 已经结束 left

    I want to interface SIM800/SIM900 or Telit GL865 to LPC2148. There is one 125kHz RFID connected to LPC2148 also. I want to upload RFID Card ID to a webserver using GSM/GPRS module. I need a FreeRTOS application for this task, which can scan RFID and upload card ID simultaneously. Card wil be scanned very fast like 70-80 cards per minute.

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