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    8,466 sip web phone windows mobile 搜到的工作,价格货币为 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

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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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    Dear Audi G 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virt...you can login via teamViwer on widows7 that the virtual machine on it and then you can login to cents via SSH or via VMware 3- there is web control also for OS and FreePBX 4- we just need to add new setting for new SIP line

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    Dear Amrit M 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virtua...can login via teamViwer on widows7 that the virtual machine on it and then you can login to cents vis SSH or via VMware 3- there is web controal also for OS and FreePBX 4- we just need to add new setting for new SIP line

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    Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you

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    Hello There i need some help to make some config on...config on my FreeBPX server like: 1-Remove and add new SIP line 2- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it note: there is teamviwer access and then you can access the Unix server by web

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    Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2-make my system cluod on internet so the agent can use there extension in anywhere 3- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it

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    Skype SIP client in PHP 4 天 left
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    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [登录来查看链接] i want to run this program on a router Router Link [登录来查看链接] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Programming Bandwith.com 1 天 left
    已验证

    Help programming sip endpoints, applications voice & sms in [登录来查看链接] I need to connect my [登录来查看链接] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [登录来查看链接] Prior experience

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    Multichannel Phone Dialer 1 天 left
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    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 8 小时 left
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    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    VOIP, SIP Expert 已经结束 left

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    SIP Proxy Server install 已经结束 left

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    ...services and chose from the ready made IVR Applications or get one developed at effective cost Click to call features: 1) When number clicked on webpage, viewer types in his/her phone number and submits to call other party, Click to call calls and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both

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    ...customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable / turn on a led to indicate a

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    sipxcom sip system setup 已经结束 left

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    VOIP ASSISTANCE 已经结束 left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    Project for Prostitov A. 已经结束 left

    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [登录来查看链接] secret GZD397XU Host [登录来查看链接] port 5060 Register String 390415541985@[登录来查看链接]:GZD397XU:390415541985@[登录来查看链接] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [登录来查看链接] secret ajFFzpNj

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    IAX SIP VOIP Softphone 已经结束 left

    ...us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application and is not

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    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    VOIP, SIP, Cyber Security 已经结束 left

    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

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    I need professional help with setting up duplicate Avaya voip phone system. I need worker with real experience. I will describe current system i have. I need to buy same equipment, and i need to know what licenses i really need to get or don't need. More details will be given as we talk together. Current System: • Avaya IP Office 500 V2. Build Version

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    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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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 [登录来查看链接] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    Predictive Dialer 已经结束 left

    ...link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list. There should be a database where agents can comment

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    Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default

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    ...looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for bot...

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    ...verification. The existing code seems to support only provisioning by email address login. Users should instead enter their mobile phone number to register. After receiving an SMS with a verification code the softphone should download the SIP provisioning data from the server. - Create IOS version and deploy to appstore - Create Android version and deploy

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    ...1main server other all asterisk server work like VM host1 IP [登录来查看链接] SIP 5060 DB NAME datastore1 host2 IP [登录来查看链接] SIP 5060 DB NAME datastore2 host3 IP [登录来查看链接] SIP 5060 DB NAME datastore3 host4 IP [登录来查看链接] SIP 5070 DB NAME datastore4 host5 IP [登录来查看链接] SIP 6060 DB NAME datastore5 i hope you understand if you can make it please

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