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    1,775 freeswitch openbts jobs found, pricing in SGD

    Install freeswitch and configure

    $56 / hr (Avg Bid)
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    7 bids

    Hello,rnI need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need your help to have a kick start of these two and then I will continue of my own.

    $11 / hr (Avg Bid)
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    Hello, I need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need your help to have a kick start of these two and then I will continue of my own.

    $13 - $27
    Sealed
    $13 - $27
    2 bids

    I need freeswitch and Kamailio initial installation and setup training. I am a voip professional but very new to these two areas, I need only few hours of your to have kick start of these two and then I will continue of my own.

    $34 (Avg Bid)
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    1 bids

    Hi Dear shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecn...shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecnical support for Freeswitch

    $120 (Avg Bid)
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    Who can build me a GSM Interception Using USRP and OpenBTS I can buy the USRP and parts on What i want from you? - tell me wich USRP i need - tell me what addons on the USRP - tell me What PC / Labtop you need - install it fully so i can intercept phonecalls and intercept SMS - maby you can build extra's like - sms advertise let me know!

    $4840 (Avg Bid)
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    Hi Dear shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecn...shweta we need website content write for our website we need use unique worlds for get more traffic on SEO ranking of website so please dont use copy and past from other website i have alredy used that . Now i want unique content words. 1-Premium Rate Numbers 2- Hosted and lisenced softswitch and billing platform 3 Voip wholesale business 4 Tecnical support for Freeswitch

    $54 (Avg Bid)
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    I have a freeswitch that is database driven using node.js I need more features to it (modification to node.js) and some amount of after support. Please state your experience with fs and node.js upon your bidding

    $27 / hr (Avg Bid)
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    Install/configure Multitenant FreeSwitch/FusioPBX + Billing ASTPP + Database PostGre + Kamalio -Servers with CENTOS 6.5 ready for installaton x 4 a) Install all the above on Centos 6.5 Servers x 4 b) Integrate FS/Fpbx with Billing c)Configure all of the above d)Test System for going live e)Training on the system x 1 hour SKILLS: FreeSwitch/FusionPBX, ASTPP billing, PostGre, PHP, JQUERY, RUBY, PYTHON

    $464 (Avg Bid)
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    Featured Urgent Sealed

    Installation ASTPP Open Source VoIP Billing for FreeSWITCH software on Debian 7 Linux server. ASTPP must be installed with FreeSWITCH on the same server Installation + voip test all the information here: The server is ready and comes with Debian 7 with SSH access

    $269 (Avg Bid)
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    We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP... * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understandin...

    $28 / hr (Avg Bid)
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    Configure FreeSwitch/fusionPBX Msql database for read/write/display from Website custom php dashboard. a))I have a Web site ready, with custom php dashboard b)I have a unconfigured FusionPBX box ready for configuring. FreeSwitch: - Extract/read/write/display user account fields/details located on FusionPBX MySQL database. Custom PHP user dashboard: - Display user account fields/details on Custom website PHP dashboard. Username/associated extension/contracted inbound virtual number/ring and pickup group/caller id number presentation option/caller waiting/follow me/Available prepaid funds sms gateway/Callback integration/ support ticket box. I need someone to: -Teach me how to implement above requirements on my FreeSwitch/FusonPBX box and Custom PHP dashb...

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    ...priorities or exceptions regarding RTP proxy; – How I can manually switch customer registrations and calls (can only update DB? Perform some action?) Comments: – Cloud PBX servers will be both Asterisk and Freeswitch; – Configuration should be designed with routing rules(i.e. route[TOSOMEWHERE] { ... } ), and they should be clearly documented (commented) Simple integration article: :realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, and I want only to add multi-server support. Don't need to integrate Asterisk DB (I have freeswitch too) to Kamailio, it must be different databases, I will sync data. Development workflow: – I will provide pre-configured accounts (4-6 accounts) on different servers and desired (Debia...

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    We need a professional for setting up and configuring an freeswitch. We want a freeswitch server which can handle up to 700 (current calls) incoming media calls (inbound, incoming calls) and for this we need someone who can configure for us the freeswitch server. The ivr of the freeswitch needs to be configure for this issue as well

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    Hello We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. ...* Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Billing and Asterisk+A2Billing servers. I'll hire best bidder with best experience and will hire on permanent basis for Technical Support as well. Full understanding of all Codecs. Thanks and Regards Must be open to integrate any cha...

    PHP
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    I have a fusionpbx and forgot the GUI passwd i need to restore it, its something quick i need urgent will adwar on the next 10 min of proyecto make public

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    I currently use Plivo for simple PHP / REST driven telephony. However, I need to have someone setup Freeswitch / Plivo () and configure it to work with a live SIP trunk that I can provide. I will also provide root access to a CentOS server. Final outcome should be that I can make and control an outbound call from a simple PHP script (similar to how Plivo works), and the call will be made using the Freeswitch/Plivo server. Need this done ASAP (within 48 hours) Thanks!

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    Need someone with experience installing a sms and voice dialer plaform. Can be an open source solution to manage our internal marketing needs. Will provide cpanel account for you to install software and setup for broadcasting. Can be freeswitch, flow route, asterisk, twilio or plivo powered. I am open to suggestions and will be sending out 100,000 to 200,000 calls a day. What is your recommendation i.e. ictdialer

    $3256 (Avg Bid)
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    Project is to build a IMSI CATCHER a GSM interceptor based on ettus research or OpenBTS (rangenetworks) Hardware : Ettus Research - USRP E110 Range Networks - Professional Development Kit *We got both hardware kits . We just need the software to extraxt imsi info . Software Features : 1. Imsi catcher (GSM/GPRS) 2. show phone number 3. MAC address If you have any question feel free to ask me !

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    We are looking for a developer who can complete some development on our Astpp billing engine which is based on freeswitch. The ideal developer needs to have strong skills in php/perl and freeswitch.

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    create DID server + ordering website I have the underlying freeswitch servers already

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    We need a professional for setting up and configuring an freeswitch. We want a freeswitch server which can handle up to 700 (current calls) incoming media calls (inbound, incoming calls) and for this we need someone who can configure for us the freeswitch server. The ivr of the freeswitch needs to be configure for this issue as well

    $533 (Avg Bid)
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    8 bids

    We need a professional for setting up and configuring an freeswitch. We want a freeswitch server which can handle up to 700 (current calls) incoming media calls (inbound, incoming calls) and for this we need someone who can configure for us the freeswitch server. The ivr of the freeswitch needs to be configure for this issue as well

    $453 (Avg Bid)
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    7 bids

    install asterisk / freeswitch; for billing install a2billing and need to modification and customization

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    Freeswitch to Asterisk will be pending and update later

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    Vbilling is a billing software for the wholesale trades that uses FreeSWITCH. We provide a 7 Debian VPS with SSH access The installation script is on the site but requires adaptation for Vbilling work properly.

    $234 (Avg Bid)
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    We require ASTPP customization. - Resellers need to be able to create their own invoice and warning templates. These templates are stored in the database. Currently ASTPP uses global templates. - We'll need to be able to create openSIPs/FreesSWITCH profile aliases so that we don't need to creat...be able to see see origination rates. These rates will need to reflect the rates that the reseller or administrator specifies. For example an administrator will import rates and add a 5% markup for resellers. The origination rates with a 5% increase will need to be displayed when the reseller views their origination rates. The same will apply to clients. Note that OpenSIPS is used as a FreeSWITCH proxy and there is a dedicated database server. All the servers are part ...

    $810 (Avg Bid)
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    I need the Astpp () billing software installed and running over "Freeswitch" () running on a server on Centos 5.5 I try but dont have the time to continue this project. Will be done remotely on a hosted server. If successful may work in further projects Any further inquire please pm me.

    $40 - $336
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    We have a Sipxecs IP-PBX that is working with an Audiocodes Mediant 1000, digital gateway. We need another IP-PBX (prefer FreeSwitch) to work with mediant 1000 for inbound and outbound call handling. Of course both IP-PBX's users should be able to communicate. This task is basically needs configuration skills. Will provide remote access to servers. More detailed information will be provided before starting the project.

    $39 / hr (Avg Bid)
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    Looking for software and hardware programmers with working knowledge of USRP software defined radio and OpenBTS with practical working skills with equipment to blast SMS in notified private area after capturing the registered cellphones in the area. Its also called as SMS advertisement system.

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    Part 1: We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We will require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We will provide remote control access to the server with the SIP line. (You are required to pick the server parts and provide a justification for the choice of parts. Please keep in mind we are hoping to scale the service quickly and the system should be able to accommodate multiple servers.) Once the system is built, we need support for a limited time and training of our technical ...

    $1947 (Avg Bid)
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    Part 1: We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We will require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We will provide remote control access to the server with the SIP line. (You are required to pick the server parts and provide a justification for the choice of parts. Please keep in mind we are hoping to scale the service quickly and the system should be able to accommodate multiple servers.) Once the system is built, we need support for a limited time and training of our technical ...

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    I need support configuring a Cisco 7970 to work on a PBX hosted on PfSense. The package installed is Freeswitch, but asterisk can be used too. The phone has been already upgraded with a sip firmwire, and the phone has been configured with a SEP Conf xnl file. At present it is unable to register with the package hosted on the PF Sense box I am very busy and could do with some help getting this to work. Its a configuration issue and need someone to talk me through the process of getting the equipment to talk to each other. On completion I will need to be clear on how i configure a System by myself. I can work with remote access and Skype to complete this task.

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    We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

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    We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. Thomas@ SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

    $1081 (Avg Bid)
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    We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. Thomas@ SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

    $538 (Avg Bid)
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    We currently are using Newfies / Freeswitch system We would like to have campaigns automatically retrieved from FTP and configured in Newfies so we don't have to do it manually. We FTP the file to a folder, system will cron to grab every 5-8 minutes. The zip will contain 2 wavs (live and answer machine versions), a csv list and , stamped with the projectID number. The list csv will be returned as (projectID)(zip). The call records will populate in the additional columns in this list before it is returned to FTP. We are NOT worried about any frontend work, as this automation will not require any frontend use. Ready to start this now. I'm online frequently

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    We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We would require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We would provide remote control access to the server with the SIP line. Once the system is built, we need support for a limited time and training of our technical staff on the built system. We are open to both hourly billing or fixed price models. Part 2: We would also require an advertiser portal where the user can log in and add advertisements to the phone call.

    $999 (Avg Bid)
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    Setup of a DebianWindows server with Freeswitch and GsmOpen. I just need the basic assistance to setup the system.

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    Hello We already have FreeSwitch Setup with Open Source VOIP billing Solution. But we need an expert on FreeSwitch with Voip billing Solution who can tune up the configurations and make our dedicated system to run high capacity wholesale voip business. We want following tasks to be done. 1. All FreeSwitch Configuration and Tunning. 2. VOIP billing features and billing must be working properly. 3. A very detail step by step and easy to understand Manual for features. It should be made in such a way that any non technical person should be able to understand it and start using billing solution instantly. It should be separately made for Admin, Sub Admin, Reseller, Sub Reseller, Users, Calling Card, Call Shop Users, Providers. 4. Complete System

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    Hello, We require the building of a VoIP Switch which...relevant SIP Account the DID is allocated to. We need to allocate a cost on each channel so each call has a cost to it based on prefix. At month end, we must be able to draw out a report stating the inbound calls via the relevant channels and its cost, with a total of it. The CORE should be something sturdy, with data saved in a database. Opensource CORE is welcome. Either Kamailio, A2B, FreeSwitch, Kazoo, or any other robust switch and billing system. There is NO outbound VoIP in this switch, its purely inbound. Ultimately, we need to know which provider is calling our DIDs, and we need to allocate a cost so we can bill them for sending calls to us. We need someone with good experience to handle this project. ...

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    I am looking for freelance who can help me to implement SIP 2 RTMP gateway with help of freeswitch. I have tried to implement it but facing some issues. I got success till: enabling mod_rtmp connecting to rtmp server with flash client provided with freeswitch source code login using username and password successfull call to 888at failure at: trying to make calls to extensions

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    Hello, We require the building of a VoIP Switch which...relevant SIP Account the DID is allocated to. We need to allocate a cost on each channel so each call has a cost to it based on prefix. At month end, we must be able to draw out a report stating the inbound calls via the relevant channels and its cost, with a total of it. The CORE should be something sturdy, with data saved in a database. Opensource CORE is welcome. Either Kamailio, A2B, FreeSwitch, Kazoo, or any other robust switch and billing system. There is NO outbound VoIP in this switch, its purely inbound. Ultimately, we need to know which provider is calling our DIDs, and we need to allocate a cost so we can bill them for sending calls to us. We need someone with good experience to handle this project. ...

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    I am looking for someone to assist in setting up load balancing using OpenSIPs, and Asterisk. FreeSwitch may also be used if it is needed. I need a load balancer and SIP proxy's and media server. The entire setup needs to be Highly Available, so that any single server can go down without any services failing permanently. I need written proposals of experience, and how many servers you would need to make this setup work. The end goal is a Multi-Tenant hosted PBX system.

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    Install, configure and document billing system Document configuration of billing system 1 hour walk through of billing system Secure server (firewall/fail2ban)

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    PLEASE DO NOT BID IF YOU HAVE NEVER DONE THIS BEFORE!! I need this implemented on Freeswitch or if you can recommend something that will be able to do about 250 calls. 1)Integrate radius with jerasoft for radius auth/billing 2)Accept inbound DID and send back IVR announcements for calling card 3)make sure correct ANI is being sent to jera , so we can do ANI (pinless) calling. 4)allow PIN entry (retry 3 times, if incorrect pin is entered) 5)allow destination number (retry 3 times) 6)Prompts in English when call arrives from DID1 and ability to give different language prompts when call arrives from different DID's . 7) if customer calls and his balance is depleted , then system must give option to enter refill voucher 8) If incorrect destinat...

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    My company is seeking ones Knowledge of Vicidial , MySQL, Ajax, Php & Bootstrap. We are running 1 large Database and 11 Agent servers. Realtime Advanced Stats to Display on a client admin panel with login prompts. Must have a proven track record and well knowledge of the exact tasks needing performed. Must have knowledge of Freeswitch on a large cluster as well. We send all traffic to our own switch and you will need to be able to retrieve the db queries from astpp db's too.

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    ...would also be useful to others. I'm surprised there isn't a site that already does this. I don't want to deal with applicants that haven't done this kind of work before. If you are going to pursue this job, please make your relevant experience clear. Also, I am hoping I can find someone that can easily accomplish far more than this simple introductory job -- someone with experience with freeswitch or equivalent, and that has experience in building call centers. Here's what I'm looking for for this simple introductory job: In the testing website you produce, the user would be able to go to the test site, and choose two things: 1) which of the 12 api speech recognition solutions they want to test, and 2) produce a recording in one of 2 w...

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    TO ONLY PEOPLE WITH EXPERIENCE WITH DONGLES : i need to setup a Voip server to use as gsm gateway for outbound calls using huawei dongles. using chan dongle for asterisk, gsm open for freeswitch, or a private module. fully manageable with web GUI with the following feature: -ability to have full control over trunks/extensions : time schedule of calls, time gap between calls, limit unique dials, maximum duration per day/hour … -ability to send call of a previously called number to the same exentsion used for the previous call, then to next dongle. -see credit per dongle based on ussd and pdu msgs and parse them correctly on the table. -send credit to one or group of dongle on gui using ussd sms and calls. -automatically detect blocked cards of dongles and disable them...

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