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    8,951 softclient sip tugasan ditemui, harga dalam USD

    Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.

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    Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will not be considered.

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    We want to integrate our twilio account with our new Obihai phone system so that any calls coming into from any of our owned twilio numbers get sent to our phone system or forwarded to other mobile phones associated with our company. We also want to have our yelp number associated with the same system. SIP Domain Sip Integration Heroku elements

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    We are looking for a developer that has used Drachtio [log masuk untuk melihat URL] You should work to make some changes to an existing Node application that uses this and Freeswitch. This will be long term relationship. Let’s start this project by creating the milestone per task.

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    I have a new server configuration and latest sipxcom build up and running for our office phones. We have two Nextiva sip trunks (xxx-xxx-9640 and xxx-xxx-9642). I have outgoing calls routing properly through the user extensions. I would like to route inbound calls from 9640 to autoattendant1, and 9642 to autoattendant2. Currently I am only able to route all incoming calls to a single destina...

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    We are looking for a developer that has used Drachtio [log masuk untuk melihat URL] You should work to make some changes to an existing Node application that uses this and Freeswitch. This will be long term relationship. Let’s start this project by creating the milestone per task.

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    Dear Coders, We are looking for a programmer who is familiar with Voip Systems and with the following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real Time C...

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    Hello, I have an asterisk PBX vers 11.22.0 . I am using a Polycom sound point IP 650. All works fine except for the transfer button. The transfer on polycom use SIP REFER to transfer the call. This is not working. Need help from anyone who know about the subject. Please respond to this project with "What up Dingo" at the beginning of your message so that I know you have read.

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    3 bida

    I need an Android app and iOS App I would like it designed and built. i Need a Sip Mobile Dialer for Android and iOS Where i will provide you my Own VPN technology where sip dialer and vpn to be merged . When Sip Dialer open vpn to be opened and when sip is closed vpn should be closed at backend . [log masuk untuk melihat URL] provide you api .. in sip it ask server ip and port insted of that ...

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    We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a pub...

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    7 bida

    Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database

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    14 bida

    Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall...

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    9 bida

    I need a whmcs addon that will provision/integrate portaone switch&billing, the idea is to control everything on whmcs without having to login to portaone. required features: 1. Client management (creation and details updating) 2. Sip account management (creation,termination, see and update call forwarding options,and statement creation) 3. SIP balance control (client to view current balance &...

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    Content Writer 1 hari left
    DISAHKAN

    Content writer required with good under standing of telecomunication network such as HFC, SIP trunk and various network device used in the HFC enviorment such as GNA, Splitter etc.

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    Asterisk with python agi 5 jam left
    DISAHKAN

    Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action

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    Looking for someone that is capable of building a cloud solution to protect from TDOS attacks. TDOS attack example is that someone is using a Script / Tool to send 500 calls per min to your phone system. The protection must occur before the call begins, from the SIP Invite, and not after the call is completed from the CDR. The cloud platform also needs to have the ability to purchase DID's...

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    As discussed. Callback request calls. We have set up an account in our support center: [log masuk untuk melihat URL] Login: [log masuk untuk melihat URL]@[log masuk untuk melihat URL] PW: IWASnrmsklah There you will also find a knowledge base with all the information. I've created 3 hours for you, because you also need time to read the documents. For the start, I have setup 15 contacts in ...

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    I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls usi...

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    SiP server with low latency and push to talk features open source implementation is preferred...no closed source and I would like documentation and source code if there is android app...documentation how setting server if using open source already existing app

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    35 bida

    Have FusionPBX installed, I need to connect with SIP

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    6 bida

    I am seeking an Outbound Proxy server to be installed on one of my servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeabl...

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    I need to link my SIP with PBX for dtmf

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    8 bida

    I started to build this web based SIP phone using [log masuk untuk melihat URL] - [log masuk untuk melihat URL] - but other work has left the project incomplete. I need the project completed and updated to use the latest version of [log masuk untuk melihat URL], 0.15.6. You will be provided with FTP access to the current source files including the HTML, CSS and current JS files. Additionally 3 S...

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    Need to list all the outlook exchange contact no's in the YEAlink phone

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    We need to configure a FreePBX server with to use 2 sip accounts. We pay 30 EUR.

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    2 bida

    Hi Ibrahim Ali M A., I noticed your an expert in VOIP and asterisk. We are having issues with our VOIP system - in particular outgoing calls through SIP trunk are getting cut off in 6 minute 39 seconds. Asrerisk server running on CentOS. Can give access through SSH.

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    I'm using a web app called SmarterTrack. It has built-in VOIP capabilities, including a softphone that has to be installed on your desktop. The desktop softphone then connects to the web interface, but I want to integrate the softphone itself into the web interface. See attachment for a rough mockup. I basically want the softphone interface to just be a dropdown box from the main navigation w...

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    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

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    20 bida

    We need a sip client app for android/iOS that could connect to a sip server and could make/receive audio/video call for a door intercom. It must have a simple interface for showering intercom video and buttons for answering/hanging/open door. Must have a config tab for entering connection parameters to sip server. Additional/optional features: Linux server app to store screenshots/videos of inter...

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    25 bida

    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

    $153 (Avg Bid)
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    2 bida

    Hi there, I need your help on the following three: #1. Adding a trunk (fyi: I was able to set up the trunk in the Micro Sip application on a Windows machine, but cannot in free PBX: Note; the username contains a "at" sign #2. Upgrade from version 14 to 15 #3. Discuss the posibilities to create a fail-over / second freepbx (as backup server) Please post per project you estimated budge...

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    7 bida

    We need the IVR with the Welcome prompt and then directly send the call agents. during the call, we have to open the database and search the request form user in the database after the end up call ticket will be open and send it via SMS to the user. Technical Need: IVR setup SIP integration SMS integration other technical support not cloud base/ on-premises base we need Database design: w...

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    9 bida

    Need to develop an windows voip application (c#.Net) with UI/UX design for SIP client Softphone which should have recording facility and integrated with our ERP using rest api detailed document is attached.

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    11 bida

    Hello, From opensource ctxphone project: [log masuk untuk melihat URL] I will provide SIP credentials for scripts/[log masuk untuk melihat URL] file. So, you need: 1. download ctxPhone project 2. Update to provided credentials 3. Make test calls making sure it works 4. Download new [log masuk untuk melihat URL] version 15.6 of library: [log masuk untuk melihat URL] 5. replace an existing 0.7 ...

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    A renowned organization is interested to work with freelance developers that have previous working experience with C / C++ and JAVA Web Development. We would be working together for a time period of six (6) months, during this time any required information (regarding the projects) shall be shared with the developer. The project we will be working on is a Call Recording Software (Compatibility with...

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    Looking for a simple SIP dialer Mobile Application for iOS and Android which can register to our asterisk server and simply make and receive SIP calls. Having G729 codec enabled is preferred, otherwise GSM codec is required.

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    I need to white-label an existing SIP dialer. (branded with our name/ logo etc..) If you have already a VoIP App developed and tested, please quote me time and cost to add our Logo & Company name. >>> Please provide references / Links to App Store for VoIP Apps you developed. App is needed for Voice off-net calling (outbound and inbound via DID) Features: - Integration with phone-bo...

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    6 bida

    I want a App Android/ IOS that can do calling others Phone Like Text With Autentification User, and Password for More informacion please Write me: [Removed by Freelancer.com Admin for offsiting - please see Section 13 of our Terms and Conditions]

    $570 (Avg Bid)
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    11 bida

    Hello, From opensource ctxphone project: [log masuk untuk melihat URL] I will provide SIP credentials for scripts/[log masuk untuk melihat URL] file. So, you need: 1. download ctxPhone project 2. Update to provided credentials 3. Make test calls making sure it works 4. Download new [log masuk untuk melihat URL] version 15.6 of library: [log masuk untuk melihat URL] 5. replace an existing 0.7 ...

    $192 (Avg Bid)
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    18 bida

    Need immediate help to configure this in salesforce. Do not reply if you have not worked with the GetHub open-cti-demo-adapter to register SIP phone to make calls from salesforce. You will be working with my programmer and this task must be working in the next 5 hours. [log masuk untuk melihat URL] I want to get this demo working with phone [log masuk untuk melihat URL]

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    Hi I have small telco using bicomsystems PBXWare. I would like to get a whmcs module to integrate with pbxware. Would like the following: 1. PBXWare WHMCS Module • Create, Suspend /Un-suspend, & terminate account • Create SIP Extension (in Asterisk) • Use Customer phone number as CallerID • Send Welcome Email with SIP account detail 2. Module configurations: ...

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    Customize a Linphone Project Description: 1. Brand Icon and Company Name on the application 2. Add account with domains in a whitelist Basic option will be used PBX - Enter your PBX Name Username - Enter your Username Password - Enter your password 3. Call logs will be only numbers - not the whole sip URI 4. In settings, G711a,G711u, G729 will be enabled by default 5. Help : Link to our web s...

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    8 bida

    I have a legacy Elastix PBX that has issues. Whenever I try to connect to the web interface it redirects me to https://MyPBXAddress/config.php. and gives me the following message You are not authorized to access this page. Enable direct access (Non-embedded) to FreePBX® in "Security >> Advanced Security Settings" menu. (FreePBX® is a Registered Trademark of Schmooze Com, ...

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    Newsletter template Design Platform: Mailjet and ERPNEXT Project: 5 newsletter designs Images: shared Pattern: A tempalte with text area is required In all newsletter should add office and show room address and [log masuk untuk melihat URL] and anythink elese so newsletter will not go to junk email . Newsletter 1 The design will be used for email marketting for the brand [log masuk untuk me...

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    We have multiple servers and I need to restrict access from general public. I would like to route all our SSH traffic, HTTP/HTTPS, Port 3306 for remote connection with MySQL and Port 5060 for SIP clients through a VPN server. Each client, depending on his credentials, will have access to some or all the servers. Also, I need to route HTTP/HTTPS of some websites through the VPN. These servers coul...

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    Server B is interconnected with one voip provider ip2ip when we send calls from thats server to voip provider ip its go through Now i have server A i want send calls from server A to server B and from B that all calls which is coming server A ip forward to voipprovider ip Server A simple will create [log masuk untuk melihat URL] trunk and when all calls dialed from that trunk goes to server...

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    Create for us a functional nice and good looking web page. - Include should be a paymaster - Should be made for Affiliate-Program - Languages DE/ENG/FR - Price plan - Chat and E-Mail configuration - Login for our members where they can see how much they are paying and their company profile - Some user interface for OUR employees which they can make some call over some SIP....VoiP - Search butto...

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    42 bida

    Hello ... i need some work in Asterisk and sip proxy .

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    1 bida

    Hi! We need to collect a statistics of messages you hear when deal a specific US toll-free number. There are two possible options so you should return us a file with 30 lines in the format: Attempt #1 - Message played "Welcome to..." Attempt #2 - Message played "Press 1 to continue this call in English" Please note you can not use any type of Internet phone (SIP, Skype, VoIP,...

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    We need a person to help us with Vicidial/asterisk configuration. We want to upload a list, do automatic calls which will say: "Say yes to connect to an agent". If the person says "YES" the call is forwarded to agent's phone (through SIP/asterisk)

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