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    10,447 softclient sip iş bulundu, ücretlendirmeleri EUR

    PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak uydurabi...

    €24 (Avg Bid)
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    SIP Soft Phone Bitti left

    Gelen Voip Çağrıları Cevaplayarak Ses mesajını dinleyip kayıt yapacak ve gelen mesaj karşısında cevap dinletecek. Eş zamanlı 500+ üzeri çağrıyı cevaplayabilmeli C# Dilin de winform uygulama olacak kaynak kod için teklif istiyorum

    €814 (Avg Bid)
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    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 paylaşırsan daha ayrıntılı bir şekilde konuşabiliriz. İyi çalışmalar. NOT: Bütçe miktarı ve süresi örnek olarak dolduruldu. Mehmet ARSLAN Computer Engineer| Manager | Co-Founder mhmttaa@

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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 yönlendirilmesi.

    €532 (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 yönlendirilmesi.

    €1220 (Avg Bid)
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    WebRTC SIP (Move from to JSSIP) Using Typescript ... i duplicated issue on ctxsip so i think jssip might be a better package as its maintained.

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    I have freepbx already installed and goip4 gateway already installed and configured. I want to configur freepbx to connect with goip4 gateway 3 lines (simcards) 3 SIP user : user 600 recive call and working with line 1 from number start with 06 user 500 recive call working with line 2 from number start with 05 user 700 recive call working with line 3 from number strat with 07 variables if user 600 dont respond he redirected to voicecall to leave a message 1 after that send sms offer 1 if user 500 dont respond he redirected to voicecall to leave a message 2 after that send sms offer 2 if user 700 dont respond he redirected to voicecall to leave a message 3 after that send sms offer 3 ps dont change any network config on goip4 gateway.

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    SIP with any RTP 2 gün left
    ONAYLI

    i have asterisk sip server and its working fine with any RTP i set on my system interface. but in same VM if i copy and make a new server when i try to set any new RTP its not working i dont know whats the issue audio is gone. so i need you to any how create a system where i can set any rtp as i want. like or or anything as i want. (we dont use our public IP as RTP we use any IP on our RTP like a eth0:1 interface ip is:1.1.1.1 so i will use this as a RTP) you can use any sip server or anything as you want. i just want to use RTP thats it. you can to setup this your local system or if you want i can give you server dont ask me any payment before test. if you can show me its working and audio is fine you will get payment with bonus.

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    Hello, we have a project need deploy Asterisk 1.8 from source code with support ODBC for user managment MySQL, and CDR record storing in MySQL, need support call recording in MP3 format and storing in with CDR details also need support video calls. * We don't allow in this project used software like issabel. * We need little bit documentate config...from source code with support ODBC for user managment MySQL, and CDR record storing in MySQL, need support call recording in MP3 format and storing in with CDR details also need support video calls. * We don't allow in this project used software like issabel. * We need little bit documentate configuration of server in small text file with procedure of deployment * We need test all with LinPhone SIP Client . Server based...

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    ...container, ready to be deployed to a virtual machine. 2. The PBX installation must additionally contain: a) browser call (webrtc) with API callback to update the status and information about the call on a third-party resource b) recording API calls with a callback to update the RECORDING URL on a third-party resource c) connect multiple phone numbers and users 3. The PBX must be configured on the SIP provider XXXXXXXX page with a functional browser call (webrtc) plugin written in JQuery, which is able to accept a call, initiate a call, put a call on hold, continue the call, reconnect to the call for a specific user 5. API to initiate browser call (webrtc) via PHP/NODEJS without having to enter a phone number on the HTML page (which works with the JQuery plugin) 6. Docume...

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    Hello, I need an expert to help me figure some things out on Mac OS. We have our own Soft-Phone that is VoIP/SIP and would like to use it to answer calls on a Mac. Is this even possible? See the image, now when a call comes in I get the Accept / Decline pop up to answer the call on my Mac but I want to use our Soft-Phone instead of Apples. Please let me know if this is possible and if so your bid to complete this project, our app is not in React.js and compiled in Electron. Thank you!

    €146 (Avg Bid)
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    I have decrypted the configuration file from my ISP's Huawei modem that contains the information needed (I hope) to configure a Grandstream ATA HT814 to use instead of the ISP modem for SIP phone service. I will provide the configuration file unencrypted. The work is to provide me with the correct data to insert into the ATA configuration so that the I can use the ATA. Thanks!

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    ...контейнера docker, готовый к развертыванию на виртуальной машине. 2. Установка PBX должна содержать дополнительно: a) browser call (webrtc) с API коллбеком для обновления статуса и информации о звонке на стороннем ресурсе б) запись звонков API коллбеком для обновления URL записи на стороннем ресурсе в) подключение нескольких телефонных номеров и пользователей 3. PBX должен быть настроен на SIP провайдера ХХХХХХХ страница с функциональным плагином browser call (webrtc), написанным на Jquery, который способен принимать вызов, инициировать вызов, ставить вызов на удержание, продолжать вызов, повторно подключаться к вызову для определенного пользователя 5. API для инициирования browser call (webrtc) через PHP/NODEJS без необходимости ввода номера телефона на HTML странице (ко...

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    Hi Kaushal D., I am looking for a Android developer with experience in firmware, VoIP & SIP. Would you be interested? Rate we can discuss over chat.

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    Calls will be routed via SIP to a server. The server will answer with messages we can upload. Various clients can upload their messages and see how many calls are being received. This will be via live and historical and summary stats. It is effectively this system: admin W6PnaGdqxNud Please analyse the features and come to me with a price and estimated duration to complete the project. Experienced required. Bids from inexperienced freelancers will be ignored. Thanks.

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    I am facing the problem. No one is in your session :8600051 ( connection with sip is remove) Call back agent Your session has been disabled Your session has been paused.. These thre problem I am facing at agent screen.

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    ...go-to for common colds, stomach issues, and everything in-between. And today, after countless conversations with my two children on the importance of drinking healthy beverages, they marvel at the thought that a simple flower in our backyard can be transformed into Meila’s Hibiscus Tonic. Named after my daughter Ameila a.k.a Meila, this tonic, is as bold in taste and flavorful as she is. Sip it hot or cold and it may just improve your wellness. Customers have found this caffeine-free beverage to be an immune system booster, a great source of antioxidants, aid in weight management, reduce blood sugar levels and blood pressure, and manage anxiety. At Meila’s, we strive to seek the best of the best from all over the globe. This in turn helps to support the com...

    €96 (Avg Bid)
    Garantili
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    22 girdi

    ...IMAGE... JUST AS INSTALLATION FOR HYLAFAX... THAT WAS JUST A SMALL JOB 15- 25 USDOLLARS... JUST TO GET THE HYLAFAX UP AND RUNNING.... LIKE APT-GET INSTALL HYLAFAX AND OF COURSE A SIP - CLIENT INSTALLATION OF T38MODEM AS WELL AS MAYBE LOCALLY USING A CAPI BASED ISDN CARD... I EVEN HAVE A PREFIX READY TO BE TAKING FAXES... AS YOU SHOULD EXECUTE A SCRIPT AFTER A FAX WAS RECEIVED AND PROCESS THAT RECEIVED FAX... AND BLOB IT INTO A MYSQL DATABASE... AS USUAL,,, JUST THAT I HAVE A DATABASE OF RECIEVED FAXES... WHEN I HAVE MY CISCO AS5400 READY... I HAVE THE PROJECT TO BECOME BIGGET ... AND YOU NEED TO INSTALL THE SIP CONNECTION USING EVEN ANOTHER SETUP OF INSTALLATION... THAT ARE ANOTHER 50,- USDOLLARS... TOTAL PROJECT... MAX. 100,- USD... FOR A SIMPLE HYLAFAX INSTALLATION...

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    Need a Cisco voip engineer to setup CME on ISR4331, with 2 ISR for redundancy/backup purpose with 15 phones with one or two fax line config along with Auto-attendant/voicemail capability. Configure SIP trunk and routing for the SIP circuits(dedicated fiber internet) provided by ISP.

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    ...Looking for a SIP VoIP React.js expert developer. Add outside 3rd party SIP VoIP service to our desktop client Electron JS Soft-Phone App for Mac and PC versions to download from our website. We currently have a VoIP SIP Soft-Phone that works with our PBX, it also does API calls where we data mine the phone number and show information about the caller from that phone number, call logs, SMS / MMS feature, Auto Dialer and more. What I’m trying to do is make it so we can put our soft-phone on the Mac App Store or provide a Mac & PC download so people can pay us a small monthly fee for the Soft-Phone but use their SIP credentials from their PBX or VoIP service while keeping all our features working. Or have it like a LIN Phone where anyone can pay...

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    In the following scenario we have Dante Via running on a PC which is sending audio sources to an amplifier With Dante digital audio input It is required that the audio sources are attenuated on the PC when receiving a call to a softphone that runs on pc

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    Hi i want a configure asterik on my sip phone with complete voip solution with all backend configuration its a 30 min task so i can pay only 500 to 600 inr bid only those guys which can do this in 30 min no pakistani bid only prefer indian and us

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    ...Redis (If needed) • GUI – Up to developer GUI / Back-end Features Required Carriers Trunks (Calls Out) Add Carrier LCR – Upload Rates for LCR per carrier (Code, Destination, Rate/Price) Ports & CPS Management per carrier trunk Carrier IP Management (Option to add more then 1 IP) Round Robin option Tick box for calls only to fail over in event of CPS/Ports being exceeded. No fail over for any other SIP codes. ANI Manipulation. Example: If ANI starts with 61XXXXXXXX replace with 0XXXXXXXX Ability to add multiple Rules. Routing Plans Create Routing Plans by choosing what carrier you want in each routing plan Customer Trunks (Calls in) Create Customer Assign Routing Plan CPS/Port Limits Add/Edit customer IP(s) and Tech Prefix Disable or enable Media (RTP Eng...

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    Özellikli
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    Lütfen detayları görmek için Kaydolun ya da Giriş Yapın.

    Gizlilik Anlaşması

    i want a simple code in C# to get incoming phone number for a sip / voip account i want to set sip server, user name , password and when this sip account get a incoming call pop up a windows and log a text file with this phone number. i want to management multiply sip account

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    Need to deploy Avaya 9608 phones on 3CX running Windows Server - Stuck on provisioning - Error "Empty SIp Proxy Server" on the phone

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    ...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/BOTIM executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/ calls. Functional flow 1) Calls originating will send to Whatsapp/gateway 2) Whatsapp gatewa...

    €479 (Avg Bid)
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    ...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/BOTIM executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/ calls. Functional flow 1) Calls originating will send to Whatsapp/gateway 2) Whatsapp gatewa...

    €479 (Avg Bid)
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    i have a simple asterisk set up where users can connect with sip phone and make a call, i want to temporarily change the dial plan so that users can just hear a audio that i saved in sounds when they try and make a call

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    We are building a simple business communication app for entrepreneurs. It is something similar to Google Voice. We have built most of the features except certain aspects of the call features itself. Who we are looking for: 1. You have a general understanding of voice communications (SIP call flows, WebRTC, SIP over WebSockets, and so on) 2. You have more than 6 years mobile app development experience. 3. You have more than 2 years experience with Flutter. 4. You are comfortable with the GetX package for routing and state management. 5. You are able to write platform-specific code (Kotlin and Swift) if necessary. 6. You speak English fluently. What we expect 1. Fix bugs and respond to support issues

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    ...understanding of the version control Git tool. Understanding and Strong skills in server-side programming skills - concurrency, network, socket, memory. Good knowledge of SIP. RTP protocols and stacks. Knowledge of development in Linux. Solid understanding of the SDLC Participation in open-source project layouts. Reviewing, testing and debugging codes Constructing database architecture as required Collaborating with interdepartmental teams for product release schedule and ensuring timely deliveries. Schedule product releases with internal teams. Required Skills Demonstrable experience and understanding of Erlang, OTP, SIP. Understanding of Event Loop architecture. Experience with asynchronous programming in JavaScript Some knowledge of basic frameworks such as HTML/CSS, SA...

    €48 / hr (Avg Bid)
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    Installation and maintenance of an Asterisk telephone system. Takeover of the old Asterisk system. (The old system is 15 years old). Many...etc. Provision of the data e.g. on an ASPX interface for other systems. We would like a contact in German. --- Installation und Pflege eines Asterisk Telefonsystemtems. Übernahme des alten Asterisk Systems. (Das alte System ist 15 Jahre alt). Es müssen viele Wählpläne neu geschrieben werden. Es sollen neuen Funktionen hinzukommen z.B. Chaträume für Konfenzenzen etc.. Neue SIP Telefone installieren. Intergration eines SIP Anbieters wie SipGate sowie weitere Verwendung des ISDN Adapter. Bereitstellung der Daten z.B. auf einer ASPX Schnittstelle für andere Systeme. Wir wünschen uns einen Kontakt ...

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    Hi Harold B., I have some problems with my magnusbilling installation, especially SIP code 403 chanunavail. Can you help me to solve it ?

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    Need a softphone application with ability to scan qr code for login Customise caller id Hold music button And few extra stuff

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    We allready setup fusionpbx system with TATA sip line... We just want to configure call flow like all call equally divided to tha agent. And most important call forwarding on mobile and Don't call the agent who is busy.

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    The project involves creating an outbound dialer software with IVR functionality.: • Develop a system that is capable of making bulk calls to recipients. • The system must be able to be hosted online and be able to connect to operator E1/SIP Trunk. • The system must be able to schedule phone calls. The system must have a phone book, that includes name, number and sex (male or female) fields, the most important field is the number. The ability to create a group should also be available. • The system should be able to create different Campaigns (Campaign creation). • The system should be able to create IVR Menu’s for specific campaigns. Example Campaign: Training people about financial literacy in rural areas, the course has 6 lessons which will be de...

    €735 (Avg Bid)
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    I'm looking for experienced web developer with WebRTC experience building dialing systems from scratch. We have a cluster of multi-tenant PBXs (VitalPBX) the new dialer must be integrated with. We expect this dialer to allow agents to place/receive calls, hold calls, transfer calls and any other normal call centre operation features to be implemented. We are open for the technologies you would like to use, e.g. it can easily be React.js with JSSIP library, or Vue with Sipml5 etc. It's would be also nice to stay in touch with developer for further development and some support in case we need something to know or to get be able to consultate with developer per hour/project basis. Minimum experience required: 2+ years web dev expertise with solid Javascript knowledge, WebRTC an...

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    Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards

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    Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards

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    Buenas tardes, en estos momentos tenemos un Webphone desarrollo con la tecnologia SIPML5. El webphone funciona de manera correcta y esta desarrollado en javascript con .net. El inconveniente se da que a veces se suspende la sesion del explorador y necesitamos evitar eso. Entiendo que necesitamos un experto en javascript que nos ayude a ver este punto y que haya desarrollado un webphone antes.

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    We are changing from one SIP provider to another one, and need configuration update. All the configuration is in place, and just need to change to the new one.

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    Acil
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    need only software for Cisco Autonomous AP IOS Software for two access points (Aironet 1600 model AIR-CAP1602I-E-K9 and Aironet 1700 model AIR-CAP1702I-E-K9) and SIP software for telephones cp-7900 , cp-7821 cp- 8841

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    We are working with a telecommunication vendor that exposes a phone number and does SIP forwarding into our system. We are using Asterisk today and seek to migrate from traditional IVR to connecting to our dialogue engine, exposed behind a websocket server. We are not bound to Asterisk if there are better components for receiving SIP calls and converting to websocket. Please reach out if you think you can help.

    €29 / hr (Avg Bid)
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    Right now we have 1 cloud FreePBX server (Ubuntu) at internet, there is Strongswan IPsec tunnel to the SIP provider (connected) but now we need set SIP trunk just over this IPsec, anything else at public IP has to be working...

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    Vici Dialer Bitti left

    1. Installation & configuration Vicidial 2. Predictive/Manual dial option press 1 campaign 3. Able to make 300 concurrent calls at once configuration & Firewall for ssh Inbound - Call menu - avatar DID groups user creation phone creation Music on hold only with avatar and press 1 then with firewall configration phone and user creation SIP Trunk and carrier configurations

    €221 (Avg Bid)
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    Configuration of DVG600s to send calls, and configure cloudfonica pbx to send calls to the gateway the gateway will send the call to PSTN. only 1 channel FXO. Please consider this requeriments: Cloudfonica PBX should send SIP Invite packets to Dlink Gateway. If FXO lines form PSTN are not working, calls can not go through.

    €48 (Avg Bid)
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    I am trying to add Twilio SIP to my Snom 870 Phone. Can you help?

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    Trophy icon Design Cake Box with Style Bitti left

    Hi, I would like to sell this famous apple pie called "Apple Strudel" in a unique and eye-catching box. The recipe of it has some main ingredients as Apple, Pine-nut, Raisin, Cinnamon, Honey and sip of rum and sugar powder on top. Would be cool to see on the box: - the logo of my shop in two color - a famous mountain: 3 cime di Lavaredo, that remind where i'm located so probably tourist like to get the cake box as holiday gift. Show me your stile :) in Italian the cake is called "Strudel di Mele" Box size: Length 38 cm Height 6 cm Depth 10 cm

    €30 (Avg Bid)
    Garantili
    €30
    19 girdi

    i need to configure D-Link DVG-6004S fxo to be able to dial out calls received from sip through pstn lines

    €57 (Avg Bid)
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