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    232 portaone openser jobs found, pricing in USD

    I'm seeking an expert in SIP and VoIP to assist me with various issues within my PortaOne system. Key Requirements: - Extensive knowledge of SIP and VoIP solutions, particularly with PortaOne - Proficiency in call routing and configuration for diverse scenarios - Proven experience in troubleshooting and resolving various issues - Clear communication skills with the ability to provide detailed explanations and solutions.

    $35 / hr (Avg Bid)
    $35 / hr Avg Bid
    11 bids

    ...logo/rebrand. We are open to options here. If you already have references or work examples of a website you build in the Telco/ISP web design field please feel free to send us some links. We have an estimate to have everything live by the 1st of April if possible. We have another platform called PortaOne for our VOIP where we also would like to have some API integration. This is not a priority for now but you are more than welcome to quote on it or provide us with advice. Any experience with PortaOne VOIP API's will also be a great advantage. Our company is based in cape Town South Africa and we have clients all across the country. Please feel free to reach out to us and ask as many questions if possible. We have 3 or for websites of our competition which we can showc...

    $1073 (Avg Bid)
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    70 bids

    I would like a proposal to create a integration from my system PBX ( PortaOne) with Zoho system. The integration must be able to handle multiple PortaOne's customer and be easy to active on day basis for each one.

    $582 (Avg Bid)
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    26 bids

    VoIP provider company is using Portaone product Portabilling version MR24 as a billing system. Portabilling installation schema consists of 5 dedicated servers: Master Slave Replication server Configurator Radius machine Slave server hosted at OVH (dedicated server with 2 disks) had issue with one of the disks, which was replaced. During replacement, the second disk was moved and it got out of the raid, which was not noticed by technician who did the replacement. As a result, after booting the server from HD to get the new disk synched, server failed. Mysql stopped (probably to avoid corruption of master mysql). The mistake was found and corrected weeks later. Technician tried to copy the content of disk2, which had slave info. After copying, turned out that 2 partitions out of 3...

    $118 (Avg Bid)
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    4 bids

    VoIP provider company is using Portaone product Portabilling as a billing system (version MR24). Portabilling installation schema consists of 5 dedicated servers: Master Slave Replication server Configurator Radius machine Slave server hosted at OVH (dedicated server with 2 disks) had issue with one of the disks, which was replaced. During replacement, the second disk was moved and it got out of the raid, which was not noticed by technician who did the replacement. As a result, after booting the server from HD to get the new disk synched, server failed. Mysql stopped (probably to avoid corruption of master mysql). The mistake was found and corrected weeks later. Technician tried to copy the content of disk2, which had slave info. After copying, turned out that 2 partitions out of...

    $135 (Avg Bid)
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    7 bids

    I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and w...FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch Skills: FreeSwitch, VoIP See more: open source technologies pvt. ltd, open source technologies , ringless voicemail, professional voicemail recording, audio voicemail greetings, british voicemail recording, recorded voicemail greetings british, voicemail voice, voicemail email elastix, openser voicemail asterisk, voicemail greetings, integrate voicemail a2billing, a2billing voicemail, multiple telephone numbers voicemail, open source ajax stock charts, open source bulk sms, marketing open flash photo ...

    $1281 (Avg Bid)
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    VoIP provider company is using portaone product porta-billing as a billing solution. There is an issue with slave server (dedicated server with 2 disks hosted at OVHcloud). It was showing apache error for https interface and the server was not accessible via ssh. OVH support reported that one of disks is defective, which was changed. Afterwards the server was given back in safe mode. We have rebooted it from hard drive and services were back and new replaced disk normally should synch with the 2nd one, but after an hour the server failed. Checks show that the server does not see the new disk, which was replaced by OVH. Raid looks ok, even the status is degraded (as it does not see replaced disk and therefore can not synch). From multiple checks and attempts, OVH support declares, th...

    $172 (Avg Bid)
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    11 bids

    The following features need to be integrated as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) Call History - ...

    $480 (Avg Bid)
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    6 bids

    The following features need to be integrated as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) ...

    $100 - $300
    Featured Urgent Sealed NDA
    $100 - $300
    8 bids

    We would like to build a full e-commerce signup process integrated with our PortaSwitch () using the Portaone API () . - Client must be able to buy the service online and get ready to use in minutes. - All the switch configuration must be done using the portaone API. - Customer receives all communication by email for setup instructions - Admin backend to allow for customisation of plans available for purchase, DIDs available for purchase, phones - Links to our website woocommerce store at to allow users to buy IP phones and then have the phone provisioned with their account info on signup.

    $2617 (Avg Bid)
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    We are starting a VoIP service and need help configuring the network to function smoothly. This could be short term help or long term relationship. We do not have a full time admin at this time.

    $27 / hr (Avg Bid)
    NDA
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    2 bids

    we need someone that knows the PortaOne PortaBilling platform version MR70 . So we can set it up and customize it to our company's needs, and then training for the team to maintain it.

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    Hi Hammad Yousaf M., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I'm looking for a similar module like the one you are busy with. PortaOne and WHMCS API Integration

    $250 (Avg Bid)
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    1 bids

    I would like to build a full e-commerce integrated with my IP-PBX ( PortaOne by API) . Client must be able to buy the service online and get ready to use in minutes. All the configuration must be done using the portaone API. Example:

    $645 (Avg Bid)
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    we need someone that knows the Telinta and PortaOne PortaBilling platform version MR55 and the Telinta Virtual Office system. So we can set it up and customize it to our company's needs, and then training for the team to maintain it.

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    ...be billed through the WHMCS billing software ( ). I need to use the API for PortaOne ( ) which is what I use as my VoIP switch and billing. The following should be possible on my website through API with PortaOne: • Customer signup and creating the customer on PortaOne • Viewing of balance of the customers PortaOne account • Increase of balance of account (called pay as you go balance by us) by paying on the WHMCS website and then the account on PortaOne reflecting an increase in balance equal to the amount of the payment. • Customer viewing call records on my website (CDR information from PortaOne via API and make it accessible to the client on the website.) • Integrate Payfast API for payments with WHMCS

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    ...like to offer you my project. We can discuss any details over chat or e-mail. I would like to install Freeswitch in Amazon AWS for production use. I prefer Windows operating system, but can also use Linux. I can run it in command line but prefer FusionPBX (but not sure if it works in Windows). I am technical, but I need an advise and Freeswitch installation and configuration, etc. I currently use PortaOne rent partition, but it is too expensive. I also have Voipswitch license, but extending support is very expensive. I like to migrate to Freeswitch because it is freeware and gives me lots of flexibility. Please let me know what you can do for me, I can offer $100 to start and see how it goes. I am new to Freelancer, never used it before. I can pay you $100 as a retainer and use y...

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    ...(2)(5) Real Time Protocol & Real Time Control Protocol. (3) Programming Languages: (3)(1) C/C++. (3)(2) Java. (3)(3) Swift/Cocoa/Cocoa Touch. (3)(4) Linux Debian Application Programming Interface. (3)(5) Android APP Development. (3)(6) iOS APP Development. (3)(7) HTML & JavaScript & MySQL & PHP. (4) OpenSource Softwares: (4)(1) GnuGK. (4)(2) Free SIP APP's( Android, iOS ) Source Codes. (4)(3) OpenSER. (4)(4) Asterisk. (4)(5) Speech To Text. (4)(6) WordPress or Any Other CMS Software. If you feel that you're familiar with All these then this project is for you. This is what we want from you. We are looking for a programmer can make for us a system to route/forward incoming calls to other phone numbers based on the extention clicked by the user. The c...

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    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 & to top up and use whmcs existing payment options to pay) 4. Password management ( view existing sip password and update sip account password) 5. Sip server management ( select server to allocate each new account to dynamically based on the number availability, since i have 2 service providers i work with) Day to day work flow: Clients add product on whmcs >> select desir...

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    ...existing team and partners for reliabl=ility and speedy setup but competitional power in the global market. (find many sources in Sales and Marketing and opensource and licensing partners and developer resource for our projects) (1) Configuration Managing Engineer of Softswitch as below; #1. PortaOne Softswitch. #2. Others (MERA , SysMaster, VPS, Asterisk, etc) (2) Both available engineer who can do R&D and Install and Business PM. #1. Quick Installation supporter for PortaOne (I will get Hosting Switch soon) = Quickstart purpose for initial Service. = SIP Phone between User to User and Calling Card from User to outside Phone#. = Must link Termination Companies' Trunk through SIP to SIP link. ...

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    1- To configure the open source Linphone Flexisip to interact with my voip server l. 2- Customize the app to interact with my PortaOne server and implementing VoipPush Notification 3 - Do necessary edits on existing codes for above adaption 4 - Successful Test 5 - Upload the fully working app including voip push notification implemented on to App store 6 - Provide the source code

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    5 bids

    ...allowing them to top op their accounts for DID usage and termination services. 5. Use Sip Router (Kamailio, Openser, Opensips) Load Balancing, Registrar, Routing outgoing calls by LCR for termination 6. Integrate a billing system (A2Billing or suggest other). If A2 Billing, code for email dunning or ability to download counts receivable (late payments) to an email platform with Dunning capabilities. Ideally, I would prefer to add dunning to A2 Billing. 7. Allow scalability 8. Integrate with CRM(Vtiger, SugarCRM or Other CRM of choice. Im open to your suggestions of other opensource packages) 9. Integrate with Mailchimp() server 10. Sip Routers (Kamailio, Openser or OpenSips ) Split for incoming and outgoing calls 11. Asterisk Realtime (4 instances) 12. MySQL Datab...

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    We are a telephony and internet service provider. We use Portaone billing system. We want to build a new signup and user area using the Portaone API and interfacing it to our wordpress-based website.

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    Install Scalable Voip Platfor...etc.) Users can select and pay for DIDs online. DID selection could be by Country, Area Code, City Code or Country and State.. The System will handle Prepaid Calling Cards allowing them to top op their accounts for DID usage and termination services. Use Sip Router (Kamailio, Openser, Opensips) Load Balancing, Registrar, Routing outgoing calls by LCR for termination Integrate a billing system (A2Billing or suggest other) Allow scalability Integrate with CRM To be installed Sip Routers (Kamailio, Openser or OpenSips ) Split for incoming and outgoing calls Asterisk Realtime (4 instances) Billing System (A2Billing or Suggested alternative) CRM (Vtiger, SugarCRM or Other CRM) MySQL Database (Master/Slave) System Testing Tools ...

    $3000 - $5000
    Featured Sealed
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    23 bids

    Looking for a female proxy help in the UI or front end interviews. There might be multiple rounds which may be phone and skype. Money will be given only after getting job for that interview. skills may include HTML, CSS, Javascript, node.js Will give $500 after selected the... css3 ui, c# developer for usa, app developer looking for a job, angularjs ui, angular developer job, urgent end year accounts help, visual web developer front end database, help paypal functionality python gae app, web developer front end skills, squid proxy help, seeking development firm availability assemble development team pick lead developer left enhance ui develop components logic ado, openser front end sip proxy, snoopy php proxy help, web developer front end designer developer karachi pakistan, video ...

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    Hi We have customized Linphone Android sdk - needs few bug fixing , If you can do it please let me know. Thanks. a) Hold button does not work during ongoing call, instead it crashes. We need to fix that so music an hold can play , which already I have activated in our Portaone sip server b) On android settings, although we don’t require normally but we can keep the proxy server option in the setting so customer can manually use the Flxisip proxy server if required. At this time , on current cofig, that optioned is removed c) Our App uses Linphone Chat server – for some reason presence indication to see if the other party is offline or online does not work in the contact or chat area. Possibly need xmpp activation. d) Fine Tune so that it can work on low bandwi...

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    To configure the open source Linphone Flexisip to interact with our Linphone IOS mobile app so voip push notifaiction can wake up app, to receive call . Accounts handled by Portaone Voip Switch . Flexisip config done, voip push certificate in place, still some issues not allowing voip push. Trying with another programmer as well, but still we are not quite there. Do necessary edits on existing codes for above adaption 4 - Successful Test 5 - Upload the fully working app including voip push notification implemented on to App store 6 - Provide the source code TOTAL AUD 100 1

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    ...l. 2- Customize the app to interact with my PortaOne server and implementing VoipPush Notification 3 - Do necessary edits on existing codes for above adaption 4 - Successful Test 5 - Upload the fully working app including voip push notification implemented on to App store 6 - Provide the source code TOTAL AUD 125. To configure the open source Linphone Flexisip to interact with our Linphone IOS mobile app so voip push notifaiction can wake up app, to receive call . Accounts handled by Portaone Voip Switch We have customised Linphone IOS sdk keeping all codes intact only changed front end to show our brand etc. Working fine, when IOS is open, but as soon it goes to foreground - it does not ring for incoming calls. We use Portaone Voip Switch, they dont...

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    Before you bid please read carefully. You must have worked with voipswitch or core4voip or portaone or vos3000 servers. Currently I have voipswitch server up and running. I need a website that will be integrated with the server. The most critical part is already done. So it will be a easy job to do. I have one web site where user can register and login. But there is no payment system. Now I'm going to give you one website what i exactly want as my site under my brand name. The site is . Please go to the website and check the functionality of it. Register and log in. Have a look their payment system. Their server is voipswitch as well. I need a front page like them. Carbon copy of it but change the color brand name etc. You should understand

    $339 (Avg Bid)
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    Hi we need a whmcs and portaone guru to integrate portaone in whms If u are a experienced developer u will know what our requirements are, but to give u an idea: account creation after payment, account suspension if needed sip accounts settings and such

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    we need a portaone switch expert to work as a noc. Only bid if you have such experience

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    Looking to build an Android app for our VOIP service. Our service is built on PortaOne. API will be provided. Required Features- *Dialpad (Integration back to deskphone or VOIP calling) *Colleague Status Panel (Shows all company users and their status, including who on the phone with and for how long) *Call Logs Screen *Click to Call all Web Phone Numbers *Incoming Call Pop-up (Answer on computer, Answer on Deskphone) *DND Option

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    what i need is the following: Becouse my telco company is growing up with more than thousands customers we need to understand on how to create a scalable system. The scenario should be: more than 1 asterisk server (I can add how many I need) load balance system and A2billing c...system and A2billing centralized engine. The load balancer must accept all the registration from the users and finally balance it on the asterisk array server. The preferred operating system is CentOS. I will pay who will prepare for me a fully functional document (doc or pdf) on how to implement that with the final step example of the file configuration. Starting from load balancer (opensips/openser), asterisk servers and a2billing setup and all we need for it like asterisk realtime config and so on. ...

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    ...communication platform which will serve as basis for end-to-end communication solution. With the right skill set and attitude you will join a young and motivated international team, working on the design and implementation of the opensource based server infrastructure. Your main responsibilities will include: * Design and development of a VOIP/SIP module within existing opensource SIP Server such as OpenSER, Asterisk, Freeswitch, etc... * Debug and analyse functional and performance issues within existing VOIP communication platform. * You will be challenged with identifying performance and scalability bottlenecks * Analyzing communication protocols like SIP, RTP, JSON. * You will also work with the rest of the team to ensure compatibility between the different server c...

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    I am opening a voip company for residential customers and i need someone to make me a step by step process for my employees to be able to troubleshoot any problems my custo...customers can have. what i need is proccess sheets /guides with step by step instructions for my employees how to handle certain problems. for example if a customer calls in he has ECHO, what are the questions (in order) that the tech needs to ask in order to be able to figure out problem, and then the steps how to fix it. p.s. I am going to use the grandstream HT502 adapter , and the portaone switch with the ON NAT feature, so that so that if customer is behind a firewall it should go through a proxy. so basically i should not be having firewall issues , so i only need process for the other possible...

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    We have a few Inter-tel 8622 IP Phones we need to connect and use with a PortaOne IP Voice Switch. As of now we have the phones successfully registered with the SIP Server and can receive inbound calls. We are not able to make outbound calls. When calling out we receive a "destination unreachable" message on the display and a busy signal. We have inspected the SIP server log files and the call never reaches our SIP server. Need to get this resolved soon. Thanks. Desired Skills VOIP Administration, SIP, Inter-tel, mitel

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    Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports ...Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable. 7) Dialer with any kind of Soft switch which support SIP (SER/OpenSER/Asterisk) 8) Mobile SIP dialer for OS Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 9) Full source code must be provided with proper comments on...

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    Hi, I am looking for someone who can setup a stable and easy manageable SIP server. We get call from a SIP trunk and we want to be able to also make and receive call from other trunks. It must be easy for me to create extra users/extensions, groups, DID and setup trunks. I was thinking about OpenSer/Kamailio with a webinterface. Can you help? Erwin

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    ...stores, data structure design and data manipulation languages. We currently use Relational (MS-SQL/MySql), Document stores(Mongo DB) and Key stores (Redis); • A track record of implementing efficient server side code. Peer to Peer Communications Development • An understating of network protocols TCP/IP UDP; • A working knowledge of SIP; • An appreciation of open source SIP stacks such as OpenSer, OpenSips, Kamalio; • A track record of implementing WEB RTC and similar technology. Personal Skills of the Candidate • Self-motivated and resourceful, with the proven ability to multi-task and operate successfully under tight deadlines; • Passion for staying up to date on emerging technology trends; • Strong interpersonal skills an...

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    Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports ...Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable. 7) Dialer with any kind of Soft switch which support SIP (SER/OpenSER/Asterisk) 8) Mobile SIP dialer for OS Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 9) Full source code must be provided with proper comments on...

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    hello i have kamailio asterisk already installed but need some configuration it installed from this doc :realtime:kamailio-4.0.x-asterisk-11.3.0-astdb im sure it will not take some hours from skilled developer also i don't have much budget thanks

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    hello i have openser kamailio need to be configured with asterisk and a2billing all those packages already installed it installed from this doc :realtime:kamailio-4.0.x-asterisk-11.3.0-astdb im sure it will not take some hours from skilled developer thanks

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    hello i have openser and asterisk installed and configured i need some help to be ready to use it installed from here : :realtime:kamailio-4.0.x-asterisk-11.3.0-astdb also i have openfire and a2billing need some config i have more information about project i will not pay more than 100$ for this project thanks

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    hello the packages already installed must be configured for pstn network also openfire included thank you

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    hello the packages already installed i need amailio + xmpp module + asterisk + a2billing to be configured in my server to be ready for voip and sip i want to use him for Audio Call Video Call Chat/ File Transfer IVR, Voicemail Call Plan, Billing, Voucher ,top please i want a skilled developer in this area thanks

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    hello the packages already installed i need amailio + xmpp module + asterisk + a2billing to be configured in my server to be ready for voip and sip i want to use him for 1. Audio Call 2. Video Call 3. Chat/ File Transfer 4. IVR, Voicemail 5. Call Plan, Billing, Voucher ,top please i want i skilled developer in this area i don't have much budget for this project so the low end bid will be hired and thanks

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    hello the packages already installed must be configured for pstn network also openfire included thank you

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    hello the packages already installed i need amailio + xmpp module + asterisk + a2billing to be configured in my server to be ready for voip and sip i want to use him for 1. Audio Call 2. Video Call 3. Chat/ File Transfer 4. IVR, Voicemail 5. Call Plan, Billing, Voucher ,top please i want i skilled developer in this area i don't have much budget for this project so the low end bid will be hired and thanks

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    hello the packages already installed must be configured for pstn network also openfire included thank you

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    hello the packages already installed i need amailio + xmpp module + asterisk + a2billing to be configured in my server to be ready for voip and sip i want to use him for 1. Audio Call 2. Video Call 3. Chat/ File Transfer 4. IVR, Voicemail 5. Call Plan, Billing, Voucher ,top please i want i skilled developer in this area i don't have much budget for this project so the low end bid will be hired and thanks

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