Freeswitch voicexml asterisk işler

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    13,879 freeswitch voicexml asterisk 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...

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    Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!

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    Merhaba, 40.000 Kullanıcı kapasiteli, eş zamanlı 10.000 çağrıyı kaldırabilecek Asterisk kurulumu için destek verebilecek bu konuda IP Telephony, Call Center, Voip, Network konularında tecrübeli çalışma arkadaşları arıyoruz.

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    Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

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    Yusuf Bey Merhaba, STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimiz şekilde çalışıyor olmasına rağmen dtmf detection sorunumuz var, ne yaptıysak asteriskin dtmfleri yakalamasını bir türlü başaramadık, bu konuda bize yardımcı olma şansınız var mıdır?Teşekkürler.

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    Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

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    Asterisk - Pbx Bitti left

    Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

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    Asterisk üzerinde çalışan crm yazılımını geliştirmek için proje desteği istiyoruz yapmış olduğumuz crm yazılımına data arama ve data kayıtlarının tutulduğu arayüz entegrasyonu yapılıcak.

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    SIP with any RTP 6 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 configuration of server in small text file with procedure of deployment * We need test all with LinPhone SIP Client . Server based on Linux OS Ubuntu

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    Kazoo cloud pbx 4 gün left

    Need experts on freeswitch , kazoo pbx, etc

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    Each file is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Vietnamese, you must also indicate the dialect) accor...rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Vietnamese, then it is necessary to indicate “Vietnamese (dialect name)”; c) if in addition to Romanian there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Vietnamese (dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    Users are not able to make calls to some countries currently. The calls are not even getting to our switch (provider for international calls). We need to check what is wrong e.g if it is a setting or something else that needs to be updated.

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    Kamailio with CPS 17 saat left

    Currently, Customer has a few dialers using asterisk and sending call to a single provider. Combined call can go can go beyond 40 calls per second, but provider only provide 30 CPS (Calls per seconds) We currently have 2 providers (A) and (B) We need Kamailio to first group the 30 calls and send them to Provider (A) every second, and overflow / remaining within the same second to send to Provider (B) Priority will go to (A) and than (B) callflow; 1. collect and gather all the calls from multiple dialer. 2. to control and send calls in batches, 30 cps. (user define number) 3. if call exceed 30 cps within the same second, kamailio will re-route next 30 calls to another provider (B) 4. able to do media bypass Thanks for reading

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    ...the billing portal through and they format of displaying client billing information is subject to the requirements 3) PBX portal - This is where the real work happens, building the pbx requires your knowledge in running native base code such as Golang to create api with connecting feature to the pbx system. The pbx module would be the pbx system dialplan design using asterisk and it should generate dialplan base on the asterisk code when that front end on the self service request configuration from the pbx. 4) Payment gateway - The portal must facilitate with payment gateway and it allows clients to see payment history on a click away. The historical data consist of their subscription and call usage where they can refer to instantly and this payment which we see should b...

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    need to make dialplan for realtime queue using mysql for asterisk system

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    Hi Arthur N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    Fusionpbx integrations and customisations

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    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed

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    There are 163 entries in total. Each duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Italian, you mus...following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Italian, then it is necessary to indicate “Italian (dialect name)”; c) if in addition to Romanian there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Italian (dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    ...build a proof of concept Java multithreaded standalone application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this Java application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. When an event is received from the asterisk server, the application will send the same event to all or some of the clients connected. When a command is received from any of the clients connected, the same command will be sent to the asterisk server. In other words, I need a "wrapper". This is just a p...

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    I need to setup Nagios or similar monitoring software to monitor my network. The Nagios server will be hosted in AWS and I will need to monitor systems (Windows & Linux) on the internet and from behind NAT. -servers -apps (Asterisk, Apache, MySQL etc) -Network outages I need someone who is familiar with this type of work and will provide me with better ways to accomplish this is possible.

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    you have to install freeswitch with astpp billing panel. from Astpp all uer & DID, trunk will manage. Voip call should work between PC to PC & PC to PSTN.

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    Hi Arshad N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    ONLY for skilled DEV in Asterisk n Freepbx ######## Help in FreePBX/Asterisk CDR REPORTS Please check ScreenShots (ivr, logs, cdr) ######## Im realy BLIND in CDR reports about: where pbx is calling to (what number), where caller is going to (what IVR/Destination 1 or 2) All outgoing calls from Dongle1 are pointed to IVR, when someone pickup call it goes to IVR ("Message: Please choose 1 or 2"). ######## TASKS: A. DID number on CDR report - need to know where (what number) pbx is calling to B. DESTINATION - need to know what button/destination/extension caller goes (#1 or #2) this is tricky couse AT THE MOMENT i can see in DESTINATION number 9 (when user press1) and number 10 when user press2 C. LOG UNANSWERED CALLS in CDR report ######## IMPORTANT: Im usi...

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    Wish to have an integration of an Asterisk PBX like FreePBX or something with ERP Next. We should have an incoming call popup with data being picked up from the ERP Next CRM Module for an existing number and We should have an option to save an unknown new number as a new Lead or Save to an existing customer/supplier / another contact similarly, we should be able to do a click-to dial from within ERP Next

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    We are currently looking for a new Softswitch, especially for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at c...for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating one from scratch. We prefer if the project used Freeswitch / Kamailio instead of an...

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    Özellikli
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    We are looking for an experienced python developer that will be responsible for creating a Python basked Backend API for a Telecommunications platform. You can think of it as Twilio + Vonage. This platform will be used to provide telecom functionality for our customers. We use MySQL (Tortoise ORM) as our backend database, Asterisk for our Media and RTP traffic, Kamailio for our front end Registration Servers. You will also be responsible for creating a client side API to access PBX features. This API will be used by our front end interface for customers and also will have Admin access for our internal CRM system to provision and manage customers.

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    Jak w tytule, potrzebuję zmodyfikować kod źreódłowy softu Asterisk (C / C++) żeby: - wysyłał dźwięk do zewnętrznego API (Google Api Speech-to-text) - na bieząco odczytywał rezultat z tego API i umieszczał wynik w zmiennej Asteriska - przykłady bibliotek Google API są umieszczone na Google Cloud Docs

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    I would need to Integrate Vtiger CRM 7.X and Issabel PBX. functions NEEDED on the Vtiger interface: - po...Issabel PBX. functions NEEDED on the Vtiger interface: - popup with caller information (only for the agent who answers) - click2call - History (including recording) What will I provide: - a fully working VPS - vtiger and Issabel (Asterisk) already installed on the VPS Important additiona informatins: - All the extensions MUST be PJSIP - I have a working version of the integration between vtiger and Issabel, but it doesn't work with PJSIP extensions. In case you are interested into working on this version, just let me know! - If you plan to customize the Asterisk callflow, you have to warn me PS: For those who don't know issabel, it is a Freepbx fork. ...

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    High knowledge expert on Freeswitch telephony programmer. Deadline: 30 days

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    Lütfen detayları görmek için Kaydolun ya da Giriş Yapın.

    Özellikli Acil Mühürlü Gizlilik Anlaşması

    Hello Ervin, I would like to ask you if you may be interested in a project like this: I would need to Integrate Vtiger CRM 7.X and Issabel PBX. With the following functions: - popup on vtiger dashboard when the agent answer - click2call - History (including recording) Vtiger and Issabel are already installed and configured by us on the same VPS. Only integration would be needed Thanks in advance Andrea

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    Lütfen detayları görmek için Kaydolun ya da Giriş Yapın.

    Gizlilik Anlaşması

    I need tools to monitor FreeSwitch PBX

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    This project is about "ISP CRM" With freeradius, Asterisk and Network Monitoring Capability which Support Vendors [Mikrotik, Huawei, Cisco, Juniper and Star OS]. Required Features and Modules # Admin Control Panel (Web + Android ) * User Management * Plan/Tariff Management * Finance Management >Invoice generation and payment processing >Recurring billing engine >Prepaid billing engine >Data Cap and Top UPs >Voice billing (VoIP CDR auto processing) >Payment gateways and credit card processing (India). >Cash Desk module * Accounting software integration (QuickBooks and Tally) * Referral Management * Sales Management and Onboard Process * Support Management * Task Management * ...

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    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed Before hiring you to our long term projects, there will be a small test which needs to be cleared. Please share screenshots, once you are successful with the test. Test Use REST API, collect all Leads with below fields, from and add display it in your Django app. Username - Demo Password - Demo (For MD5 password, you can translate ‘Demo’ online) REST URL endpoint: Module name for connecting to SuiteCRM Leads Field names to collect are ‘ phone_work, first_name,

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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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    Require help regarding calling

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    Hi Abu K., I need some help to configure a phone survey on a client's Asterisk server and integrate the results with Stats () Can you help me with that?

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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 dial plans need to be rewritten. New functions are to be added, e.g. chat rooms for conferences 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...

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    I need someone who can sucessfully setup Asterisk and Google Speech to Text. More addons will be done later which is payable extra.

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    Install TLS 'let's encrypt' certificate, setup PBX to use TLS with SRTP

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    Hello Friends, I am looking for a full time VoIP telephony Developer who can create a web based SAAS Application for Call Management Portal that can help business to manage their huge call volumes. For more details, I can also show you the reference applications.

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    Hi Arpit M., I have a MAgnusBilling softswitch based on asterisk and I have some issues that I want to fix. Can you help me for this ?

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    I have an in-house build custom CRM and Asterisk for call center. When there is an incoming call, I want a pop up on on agent screen showing customer info from our CRM.

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    I have installed magnusbilling which come with Asterisk 13 on Debian Need someone to configure Webrtc clients to connect to it useing web phones And provide a " how to " guide file Thanks

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