Asterisk pbx serverPekerjaan
Memindahkan database dan script yang telah terinstall di server centos 7 yaitu menggunakan asterisk. Software telah tercustom akan dipindahkan ke server baru. Diharapakan yang telah memahami asterisk dan freepbx
asterisk untuk voip membuat sever pribadi
jasa setting untuk keperluan kami yaitu: Billing System / Budgeting per ID / User Recording All Call IVR / Auto Attendance Call Fax to Mail & Mail to Fax Bisa Cabang IP PBX Yealink BLF [Busy Lamp Field] Sip Messenger connect Android APPS VOIP saat ini kami memakai program elastix dengan hardware openvox a814hj0582xk dengan chip 8fxo
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We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone
...(inclusive) and store these into an array. Produce a chart EXACTLY like the one below that indicates how many values fell in the range 1 to 10, 11 to 20, and so on. Print one asterisk for each value entered. Notice the spacing for everything. Range # Found Chart --------- ---------- -------------------------------------------
We are looking to do the following in our FreePBX Software... 1) Someone call into the PBX System, hear an announcement/instruction to 2) Enter their Phone Number 3) The System then save the imputed phone number to a CSV file 4) This must be unique for a extension assigned to a DID Number. (can have multiple calls at same time) 5) File will should
...ou seja faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar out...
Asterisck pbx server installation (fusion or frepbx), configure and integrate with Odoo VoIP module v10 for Web based calling using WebRTC protocol. Need proven or similar project experience in Odoo platform.
Configure Asterisk URA with PHP integration
Need to install a Eslatix Server with PBX and VOIP capabilities
Necesito generar una maquina virtual con la telefonia IP asterisk, elastick u otro servidor.
...'moodle'): [login untuk melihat URL] Required fields are next, where are same fields obtained from the original report, plus (marked with asterisk*) four fields that will be calculated using the same extracted data: - Date and time - First name / SurnameSort - Email - Grade item - Original grade - Revised grade -
I need a person who performs Vtiger CRM Integration with Asterisk
We're looking for a senior React Native/Redux developer to complete the final steps in a VoIP/Text Messaging mobile app. The mobile app uses PJSip to communicate with Asterisk, and interacts with a back end API created in PHP. Ideal candidate would have knowledge of React Native, Redux, and PJSip (Optional but definitely recommended). The final stages
need an Asterisk developer to help on configure a GUI for my new VoIP infrastructure Task - design and configure my Asterisk Do not bid if you are not an experienced Asterisk developer
Hi , I am looking for someone who is able to configure ZRTP on freepbx ? and would like to know if all features work with this protocol? I read some articles said that call recording is not possible with ZRTP. Does it work with all other protocols such as, TLS, UDP and TCP? Let me know if you are interested in this kind of work and let`s discuss the duration and price.
We want a site where we easily can see the calls that come in and what happens to them. An example could be A customer calls 70209404 (NordicCall), the customer is in the queue for 2 minutes because every agent apart from two are on DND, one agent is busy an the other agent rejects the call. So we must continuously be able to see all of the information, and there is a site with further informati...
Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload
...with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat
Hello. I need to implement a click to call system on my website. I've a li...duration. He can be billed 10€ for 10 minutes or 20€ for 20 minutes. Only after the site deductes the credits, the call will start. It needs to connect to some cheap voip server (3cx, asterisk, freepbx, etc). The support tech person cannot see the client phone number.
...Admin - login access Agent - login access Customer - login access Admin - Features Add, edit, delete Customer accounts Assign customer phone numbers (integrated with Asterisk ami to enable screenpop) Add, edit, delete Employees (no login) Add Customer business info (screenpop, location info) Add screenpop Forms Add changes reason text box.
...will be the same as [login untuk melihat URL] [login untuk melihat URL] 2. I want to integrate Vtiger7 with free Pbx - This needs to be done immediately - Its a small task . I will give you access to free pbx and the Vtiger. [login untuk melihat URL] Integration with Vtiger Version 7 3. 3. I need Vicidial to Vtiger
...back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers are
Integrate Issabel-Asterisk with Twillio Elastic Sip Trunking to records Calls from Twillio (through Flex)
Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.
internet marketing to local business. our company sells technology products, like pbx, servers, up phones, security cameras and hardware and software, security information. our web site is [login untuk melihat URL] we want to have more leads in these areas. could you help us in this process.?
One Server with multiple disks managed through KVM - qcow2. All disk OS are Ubuntu 18.04. Security is important. To check the server, it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache2, Certbot - SSL, SMTP server, HTTPS, PHP, MySQL, KVM - qcow2, LibreOffice, Ubuntu, Asterisk PBX for invoice info., etc.
Hello, I need someone to install a working version of GoAutoDial v4. The instructions are here: [login untuk melihat URL] I will provide the VPS required and SSH access. Requires: PHP, MYSQL, COMPOSER, ASTERISK knowledge.
Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .
...[login untuk melihat URL] Please refer to our competitor websites for reference: 1. [login untuk melihat URL] 2. [login untuk melihat URL] 3. [login untuk melihat URL]
A2billing corrupted no sync with asterisk , unable to make outgoing calls although customer has credit
Migrate Queuemetrics 1.5.0 and asterisk 1.6 to lastest Queuemetrics version and asterisk (per best recommendations from Loway)
Need help with enabling join Q on a queuemetric 1.5.0 and asterisk 1.6. The idea is that agents can select the queue from queuemetrics agent area.
Have Zabbix installed and working. Needs fine tuning. Host include: Windows 7 through server 2016 with active directory. Printers Xerox, Canon, HP Peplink routers, Balance and SOHO series FreePBX (asterisk running PJSIP) Ethernet switches Nortel/Avaya, Arista, 3com Microsoft SQL 2016 Standard with AG (to be added) Multitech rCell 100 Cellular router
Looking out for someone who has good hands on FusionPBX, we are looking out to move over to Fusion (Freeswitch) from (Asterisk). We are in Melbourne, Australia.
VTiger Installation , Click to call & Configuring of Cepstral TTS, configure scripts, Integrating payment gateway
i have asterisk Voip FreePBX installed on my server at home and i want to let it show the caller id on all of my tv i have chromecast on all of my tv and 2 of them are smart samsung tv tell me if you have solution for this
I have a number of phpagi scripts that work on my older system, but now that i'm moving to a new asterisk system, they fail to run. They all suffer I'm sure from the same problem, whatever it is. Old system uses PHP 5.3.10 And new system uses 5.3.3. I'm not sure what else you need to know, but ask away.
Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.
Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.
Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.
One Server with multiple disks managed through KVM - qcow2. All OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache2, Certbot - SSL, SMTP servers, KVM - qcow2, MySQL, PHP, LibreOffice, Ubuntu, Asterisk PBX for invoice info., HTTPS, etc.
Asterisk devlopment for voice broadcasting and vicidial
I want to add PAI to all outgoing calls from an asterisk to a provider. Calls are from other pbx and are received without PAI. Caller CID from uri must be send as PAI. The system is Elastix 2.5
...features) to successfully deploy phone identity certificates to establish the identity of a client or server during a TLS session. Phone will also need to be added to Windows Active Directory as a computer account. Windows Server, AD forest, PBX/Communications Platform and IP phones have already been deployed. Deliverables in the project include:
...sent is normally a website URL. I have this working using vitelity sms service, but I need more robust sms handling. Mainly to handle 'stop' requests and etc. I'm using the PBX in a Flash Incredible 13.13 platform. I would like to try to nexmo first, then later add other providers. I'm sure there is more needed from me to do this project, but if you
No puedo conectar mis gateway GSM con asterisk y tampoco puede escuchar ni tampoco mi audio en las llamdas, necesito un experto en Asterisk.
We are looking full stack developers to build a telephony system based on engine Asterisk or Freeswitch. We need to build the whole system including a client area and manager interface. Experience on VoIP/Telephony proyect is a must. Please contact us for more details.
