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    8,662 softclient sip iş bulundu, ücretlendirmeleri EUR

    Hi thepasserby82 Merhaba Ankara'da bir şirketiz.,Bir VoIP-SIP programı geliştirmek istiyoruz. Farklı olarak speech to text ile entegre olacak. Bunun örnek kodları github ta mevcut. Bunu imkan veren en iyi voip open source çözümü olarak asteriks bulduk. Türkiye de biri ile çalışmak bizim için daha avantalı olacaktır. İletişim bilgilerini pay...

    €888 (Avg Bid)
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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

    €493 (Avg Bid)
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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

    €1131 (Avg Bid)
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    2 teklifler

    Currently running freeswitch ver1.6.20, would need expert's help to implement iOS push notification to trigger linphone sip client to register and ring.

    €139 (Avg Bid)
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    ...chat. I need an application that runs on Android phones (4.0 or above) and can send/receive calls 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. Full description below. Please review and message me with considerations. If any

    €888 (Avg Bid)
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    I wish to write a java application which will send a sip invite to sip server using jain sip api's . I have my code which is sending sip invie but throwing ObjectInUseException .

    €14 / hr (Avg Bid)
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    Hello, We are looking for Windows Phone and Desktop SIP application with our branding using Linphone. Do you have expertise? If yes please mention in the start of the bid what coding technology is used for Windows Phone and Desktop.

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    I have a list of UK mobile numbers in csv. These are CLIDs that need to be provid...the response 14108000111111,[07534535875],,,,3,,111111,11 [07534535875] is the CLID from the CSV and 14108000111111 is the external number to dial 1. I will provide a UK sip trunk to the winning bidder 2. This project needs to be delivered in a maximum of 3 days.

    €543 (Avg Bid)
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    SIP client app 1 gün left

    We are looking for SIP client application to allow our customers to make and receive SIP based Voice/Video Calls over Wi-Fi using any Android/IOS Devices. Our PBXs are basically Asterisk based & we have SMB & Enterprises level IP PBXs , we are looking for a good app for this , please send me if you have similer solutions for this

    €542 (Avg Bid)
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    FREEPBX EXPERT NEEDED 17 saat left
    ONAYLI

    HI, I HAVE INSTALLED FREEPBX ON MY HOME NETWORK, MY PHONE ON LOCAL NETWORK WORKS WELL, BUT NOW I HAVE PHONE TO CONNECT OUTSIDE MY NETWORK (REMOTE EXTENSION) ...HOME AND MAKE PORTFORWARD ON MY ROUTER TO MAKE REMOTE PHONE WORKING WELL I HAVE PUT THAT ON MY REMOTE PHONE : DOMAIN : [login to view URL] USER ID : [login to view URL] PASSWORD : PASSWORD SIP PORT : 5060

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    ...User A is offline and User B calls to A, then Opensips should send Push notification to User A. (Opensip Side) User A will send SIP REGISTER to Opensips. (App side) Opensips will receive the SIP Register success event and send SIP INVITE to User A. (Opensip Side) User A will ring and answer the call. (App Side) In Above flow, App side is implemented. We

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    Need an iOS SIP client based on pjsip 2.8 library. iOS client app should seamlessly work with asterisk 15.4.1 i.e should handle registration with authentication and support voice calling functionality. iOS client app should handle NAT64 as needed by Apple for app approval. Code should be in Objective C

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    The iOS/Android SIP client functionality: - connects to the SIP server through a VPN based on OpenVPN or L2TP over IPSEС - push notification - chat based on SIP/SIMPLE - encryption for voice - use scp for load setting

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    Our SIP Trunk provider recently made some changes overnight. Originally, we were configured for PjSIP, and could not register. After changing to chan_sip, we could now register, however can only make outgoing calls. When we try an incoming call, the SIP trunk provider tells us that our FreePBX instance is returning 401/unauthorized to them, and so

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    ...-Raspbx(Asterisk11 +FreePBX) with 4 chanmobile bluetooth module already working -Grandstream UCM6202 with GoIP-8 Hibertone GSM gateway. I can make and receive calls from my SIP extension to GSM networks with the gateway or with the Raspbx chanmobile. The problem is that i cant use the conference call of GSM (call 2 external numbers with one GSM channel/chip)

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    Dual Sim SmartPhone Project Objective We like to terminate calls via SIP to the phone, then making the phone's second GSM module to make a mobile call and switch these calls together. It is like a one port sim box or also called GSM gateway (see [login to view URL], [login to view URL], [login to view URL], [login to view URL] ). The user of the smartphone shall

    €537 (Avg Bid)
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    hi i need to make call from sip regist to andriod gsm network

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    Елена, привет! Вы сможете сделать приложение для андроида чтобы он мог принимать звонок через sip приложение и совершать звонок по gsm. при этом абоненты должны слышать друг-друга?

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    i wanted to develop app which can make android phone work as gsm voip gateway. goal of this peoject is to replace traditional voip gsm gateway with android phones, each phone will represent as an voip channel/port. rest you know better. some additional features will be discussed in inbox

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    Logo Design Bitti left

    I am currently setting up an entity and need a logo for the company. The company is called SIP and it is in the petrochemical industry.

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    I need to create a sip inbound and outbound call base on webrtc project [login to view URL]

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    7 teklifler

    I need an application that runs on Android phones (4.0 or above) and can send/receive calls 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

    €1066 (Avg Bid)
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    1 teklifler

    I have a list of UK mobile numbers in an excel sheet in the format e.g. 7534535...the solution to conequentively or simultaneosly dial these numbers and fill the database with the results. The solution can be a desktop script that takes advantage of a softclient or a dialer script (goautodial) ** numbers given in the above example are just examples.

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    10 teklifler

    looking for some one whos expert in 3cx setting to link with grandstream pabx sip

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    ............................... 1. For users register with their phone number (as we have in rebtel) --- it automatically pulls the number from the sim Question(Do I need an SIP provider for the video calls)? 2. For Video and SkyTelle to SkyTelle app to app calls to be completely free 3. Can we get a pretty theme to use so we can complete asap? 4

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    Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or

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    looking for some one whos expert in 3cx setting to link . with grandstream pabx sip

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    3 teklifler

    ...mysql + php Crm we need to generate reports accessing db from asterisk pbx where are call logs are stored. we need to register each click on phone number callto link activates sip client ( zoiper or eyebeam) and makes a call on asterisk (freepbx) so maybe it needs a backend api so it gets info form browser clicks and gets info from freepbx databases and

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    Hi there, I see you have placed a bid on another project to install Homer on a sip server. We currently have a SIP server (Windows Server 2012R2) with a MySQL DB and Listeners running on Linux (Ubuntu). We are in need of VoIP monitoring software to check quality of calls for troubleshooting (would like to know why clients quality is bad) and if

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    Setup Homer SIP monitoring of high traffic VOIP cluster

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    SIP Messages Bitti left

    SIP stands for 'Session Initiation Protocol'. We need to send xml messages via SIP Messages. The SIP wrap the xml. we need to send messages from our server to another server, and to recieve messages as well. no need to implement voice related things, only simple data. ptobably need to use one of the existing SIP clients sdk/open-source, and then it

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    ...This server cannot register to a BSNL SIP service that originates in India. The purpose of the project is: 1) set up a SIP proxy on a server in Bangalore India (machine provided). 2) Register SIP proxy to BSNL. 3) Connect this to the USA server so that it can make inbound and outbound calls to India using this SIP proxy, to achieve the existing functionality

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    Looking for Twilio expert with expertise in receiving inbound calls into Twilio that also contain call data. This might be via SIP transfers with details in the Call Header but would also like to learn other ways to achieve this. Knowledge of SIPs, MLPs, and Flashhook transfers would be helpful. This does not require travel. This will be discussions

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    We want freelancer who already work in jain sip API

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    ...above) and can receive calls 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. Features - Route call from SIP to GSM - Be able to dial the recipient number carried in the SIP call - Convert audio from SIP to GSM networks - Able to run on Andro...

    €151 (Avg Bid)
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    I want to make a Huawei Dongles to be GSM gateway connected to PBX for making calls. A USB hub with 5 ports will be connected to computer so that ther can be 5 dongles in use simultaneously. The purpose for this project is to facilitate voip call termination in huge volume.

    €405 (Avg Bid)
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    I want to make a Huawei Dongles to be GSM gateway connected to PBX for making calls. A USB hub with 5 ports will be connected to computer so that ther can be 5 dongles in use simultaneously. The purpose for this project is to facilitate voip call termination in huge volume.

    €418 (Avg Bid)
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    ...for android phones. The purpose of the app (or other alternative) is to serve SIP ----> GSM gateway. Calling should be automatic - No human intervention. The app should be capable of working continously 24/7 dialing out calls as they are received. The phone will receive the SIP/VOIP call through the data cable while normal GSM SIM card will re-originate

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    SIP SIGNAL Bitti left

    please read carefully before accept my job I have termination gateways work with CDMA network not GSM have issue that is when make call it run billing directly with calling...billing directly with calling before ringing or real connected this happen even i use my mobile for same network so is any one can fix by soft switch (Signal billing inside sip)

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    Setup Vicidial for an inbound / outbound contact center on IBM Baremetal with Virtuozzo OS and Twilio Elastic Sip trunk

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    ...locuciones. Reproducir una lista de locuciones y poder capturar DTMF Poder transferir la llamada (añadiendo y modificando algunas cabeceras SIP) Poder liberar la llamada (añadiendo y modificando algunas cabeceras SIP) Grabar una llamada. Grabar un mensaje Revisa la documentación adjunta, para ver el protocolo en detalle...

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    ...need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server

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    ...User A is offline and User B calls to A, then Opensips should send Push notification to User A. (Opensip Side) User A will send SIP REGISTER to Opensips. (App side) Opensips will receive the SIP Register success event and send SIP INVITE to User A. (Opensip Side) User A will ring and answer the call. (App Side) In Above flow, App side is implemented. We

    €107 (Avg Bid)
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    1 teklifler

    ...Claim/Disputes management  Service Improvement Plan: o Clarification, improvement and alignment of processes within account. o Preparation/updating of SIP o Steering of SIP meetings, leading of SIP program with internal and external counterparts. o Responsible for implementation of improvement actions (leading role) o Is developer and promoter of best

    €99 / hr (Avg Bid)
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    ...to my site ask a question | pick a fee | then receive their custom answers or consultation. I would like my site to be very closely designed like [login to view URL]|ga|26|ALL%20-%20Medical%20-%20General%20Medical%20-%20Search%20-%20Google%20-%20Exact|ALL%20-%20Medical%20-%20General%20Medical%20-%20Online%20Doctors%20-%20Exa

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    HI, I HAVE INSTALLED FREEPBX ON MY HOME NETWORK, MY PHONE ON LOCAL NETWORK WORKS WELL, BUT NOW I HAVE PHONE TO CONNECT OUTSIDE MY NETWORK (REMOTE EXTENSION) ...HOME AND MAKE PORTFORWARD ON MY ROUTER TO MAKE REMOTE PHONE WORKING WELL I HAVE PUT THAT ON MY REMOTE PHONE : DOMAIN : [login to view URL] USER ID : [login to view URL] PASSWORD : PASSWORD SIP PORT : 5060

    €25 / hr (Avg Bid)
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    Hi im looking for someone who can develop or customization SIP firmware for Avaya 1608 IP Deskphone. Please note that Avaya does not have SIP firmware for Avaya 1608 IP Phones. I've done it before or if you can deliver the job you are most welcome

    €185 (Avg Bid)
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    ...number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk,

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    ...forwarding ) Softswitch in the USA now. = We buy one server from DIDx with Worldwide DID connection service through API. = Now 5 US DID Originations to PSTN or SIP to worldwide. = We can add worldwide DID on this Server immediately. I am looking for expertise in Telecom fields (1) Programming Developers and (2) Business PM who

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    SIP voice voip Bitti left

    1-You are able to connect it to twillio 2- Have the source code for a simple SIP phone 3- Able to display any number when calling out. This is a pretty quick job.. I'm seeking for someone already familiar with this

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