Freeswitch openbtspráce

Please find requirement below: Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones

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

We are a small IT-service company which provides besides general IT-services specific products. These products are build on linux and need developement and maintenance. One of this products is a virtual multitenant PBX based on freeswitch with fusionPBX, everything completely opensource. Your job would be to maintain the PBX, be in contact with the developers of the software and client-activation. Furthermore would you manage another product based on "PYDIO" and general assistance in our Linux environment. Knowledge in Microsoft environment is appreciated, since third party software connected to the PBX is running on windows server. Strong understanding of general networking (routing, switching,...) and virtualization is absolutely required. Right now we are 2 pe...

I have a freeswitch install and runs a custom lua scripts to direct call flow. Currently the system accepts callers input then plays a confirmation message and once confirmed a postgreSQL query pulls the required data and places it in session. What I need is a parallel process (thread) that collects the digits and immediately queries the DB while the caller confirms their digit entry. If the caller confirms their digits then the data is already waiting in session. I have several other projects but will use this project to qualify the freelancer.

We are looking for a developer who can assist us with custom changes to our astpp billing engine which is hosted on freeswitch, the project is already 80% done we just need someone who can complete it

Server: Asterisk, Kamailio, or FreesWitch on Amazon EC2 - SIP protocol - File transfer (photo, video, anything, ...) - Animated emoji, stickers and ringtone - Platform: iOS, Android, PC, Windows Phone, Mac, BB, Nokia, and Kindle. - AnyFirewall: STUN/TURN/ICE - Cross-platform code. - SIP interconnection for PSTN termination - Design UI - UI impelemation for all above platform - Cheap International phone voice call (live Realtime A2Billing) - API for Paypal, credit card payment - Report analytic. - Data-base: MySQL, PostgreSQL - Radius client for accounting. - Live monitoring ASR, ACD, call-statistics by destination gatewa - SNMP statistics - LCR Routing - ASR Routing - Average Call Duration (ACD) - Gateway/route load - Route capacity - Number of calli...

I need freeswitch configured and integrated to Jerasoft billing to accept inbound DID CALLS and return full IVR prompts for calling card. Freeswitch must send radius request to jerasoft billing, so that customers can get balance amount, durations etc. Please PM and we can discuss in detail this project .

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

K zobrazení podrobností se zaregistrujte nebo přihlaste.

We need similiar functionality as Mod_Avmd does in freeswitch, please visit following link for detail

Hi We want to make V5.2 to SIP Gateway. V5.2 is a Telecommunication protocol for connecting Access Network(ONUs) to TDM Local Exchanges with E1. It is preferable to add V5.2 project to one of Freeswitch or Yate or Asterisk. V5.2 implementation on both ONU side and LX side are required.

Hello, Dear Freelancers We need a bandwidth optimizer for Freeswitch. we have customized freeswitch with billing system the problem is whenever the client placing a call using G729 codec the bandwidth for that call is 24kbps. do we have any solution for that like RBC (Reduce Bandwidth Consumption) or optimize bandwidth for G729 codec by removing unwanted RTP header or payload without any voice quality please bid if you know about my project. Reduce Bandwidth Consumption

We would like to have a Kamailio and Freeswitch training intermediate and advanced level Training goal is to be able to understand the following: • SIP and IAX protocols • Kamailio structure and main setup and config files • Kamailio load balancing • Freeswitch dialplan syntax and constructs, • Connecting Freeswitch to a VoIP provider • Call monitoring and CDR • Effective use of Freeswitch features • Pointing DID to Extension and Call routing • Billing with freeswitch • Connecting Freeswitch to a database • Troubleshooting your Freeswitch system We look for someone with background in VOIP and have good communication in English or French Thank you MJ

We require someone experienced in VOIP, specifically OpenSIPS, MediaProxy and Freeswitch to aid in the implementation of a class 5 switch solution. Proposed design; * Two OpenSIPS proxies - All customer termination and origination from / to customer via proxies * Two MediaProxy media servers - All customer RTP is between MediaProxy and Freeswitch * Two Freeswitch SBC's - Acting as PSTN gateways between OpenSIPS and carriers / TDM gateways * OpenSIPS Location / Redirect Server * FreeRadius Server - AAA Proposed design attached as a PDF. Requirements; * Customers can either Register (Dynamic) or trunk directly (Static - IP Auth) with Proxies * Customer Location / AOR needs to be maintained by Location / Redirect server. When call received from carrie...

Project Description: Project Description: Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server Freeswitch to Client using G729 codec. Bid and come online for more discussion

...balancing and redundancy. Also to have an astpp package setup with sample customers and the ability to do monthly billing through my credit card processor at a set monthly rate. It is very similar to the document "Load Balancing of network" that is posted in your portfolio at this site. As of now I believe that I would have 1 opensips server and 3 freeswitch/fusionpbx or Kazoo servers with a common database on one of the freeswitch media server. I would have the servers setup and running CentOS 6.5 with a shell account for you to access. I was wondering if you could give me an estimate of the cost for a project of this nature. I am also curious what cost would be for ongoing periodic installation and configuration of newly added systems and systems ...

Build VOIP PROVIDER platform SoftSwitch on Asterisk/FreePBX, FreeSwitch, MOR X4 Class 4/5 softswitch with billing and routing, SIP proxy. Example reference 1) Example reference 1) End product to be similar to and with all features as in: , REQUIREMENT A) Our website and customer portal to be connected to backend and final front and backend must look similar to REQUIREMENT B) Required backend Features: - Scalability to up to 10 000 concurrent calls, Realtime pre/postpaid billing, Virtual numbers, webportal, - mulitpoint registration, internet failover protection, call transfer, music on hold, Audio and Video conference

I want to have following. Please quote ONLY if you KNOW what i am talking about. Please describe Components you shall use before i can shortlist. You might be asked to manage it permanently if found suitable. it shall be a large long time revenue generator A) SIP Proxy with Asterisk cluster and/or Freeswitch Cluster or freepbx cluster B) Multi tenant each with Multiple simultaneous conferences having Video+ Audio. C) IVR for virtual PBX with a front end to add IVR menus D) Audio/ Video conference between multiple clients. Multipel such conferences should be possible. E) Billing Menu G) Web rtc to SIP server- Audio/Video conference

We are looking for Freeswitch development. We have an existing IVR application which we want to convert to Freeswitch. Please share your expertise in it and sample application to evaluate your experience in this domain.

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.

Configure an freeswitch and elastix servers as follows: Freeswitch server as follows: • Configure sangoma Quad T1 board: o Four ISDN-PRI, protocol 5ess, ESF, B8ZS. o Two will receive inbound traffic with numbers to be dialed out on the other two American telephone Format with 9 for outside line (9-XXX-XXX-XXXX) o The other two ISDN-PRI are connected to the Central Office and receive I outbound and inbound traffic. o PRIs pass ANI and DNIS information to and from all four PRIs • Configure server to record all calls that go in or out of the server. Recordings of call, if possible, will have file names that contains number dialed outbound/Number called Inbound and stored in directories by date. So directories names will have date MMDDYYYY, and file names will ...

I have and existing cloud based freeswitch/fusionpbx install- I need someone to fix a few bugs and improve on the code. I'm looking for an ongoing administrator that has time to track down and repair issues. The site is in development but nearing completion. I will pay you hourly and depending on your skill set I need an ongoing administrator.

Simply put, I would like someone to use the following link to put together a federated VoIP system on Amazon AWS. The system should be able to support 10,000 users per the federated VoIP system spec. I will provide the AWS and my VoIP provider trunk info. http:/... To deem the project closed, there will be 3 conditions: 1) Use the Lumicall VoIP client () on an Android phone to call a regular POTS phone. 2) Use the Lumicall VoIP client on the Android to call another Lumicall VoIP client. 3) Use the POTS line to call the Lumicall VoIP client. Rather than Asterisk, I'd like to use FreeSWITCH. I will provide additional info, but this is the nuts and bolts of what is needed. Please ask questions if you have any.

I have and existing cloud based freeswitch/fusionpbx install- I need someone to fix a few bugs and improve on the code. I'm looking for an ongoing administrator that has time to track down and repair issues. The site is in development but nearing completion. I will pay you hourly and depending on your skill set I need an ongoing administrator.

I have and existing cloud based freeswitch/fusionpbx install- I need someone to fix a few bugs and improve on the code. I'm looking for an ongoing administrator that has time to track down and repair issues. The site is in development but nearing completion. I will pay you hourly and depending on your skill set I need an ongoing administrator.

I would like to integrate ASTPP within my existing Freeswitch + FusionPBX. FusionPBX is installed on a nginx webserver with a mySQL DB. I would like to integrate ASTPP also with the nginx webserver and another MySQL instance. Could I have a document with the steps you took to integrate ASTPP within the existing environment. In case of a crash of failure, I would like to setup the environment on my own.

I would like to integrate ASTPP within my existing Freeswitch + FusionPBX. FusionPBX is installed on a nginx webserver with a mySQL DB. I would like to integrate ASTPP also with the nginx webserver and another MySQL instance. Could I have a document with the steps you took to integrate ASTPP within the existing environment. In case of a crash of failure, I would like to setup the environment on my own.

Need training on FreeSWITCH, FusionPBX and ASTPP with extensions. Dial plan other related features and programming handset with my own network. I will be using Synapeoglobal to host my voip servers. Need training on Basic trouble shooting and server maintenance. installation of ASTPP and compiling with FreePBX. Configuration, troubleshooting and Back up. This is the new project to set up a International Calling card and Also Residential Land line and Business Voip Provider. Need to install and configure for the practice and make it work at the end of the training and the live installation will be done on VPS to make it go live. I have a very limited budget.

помощь в установке и конфигурации server freeswitch

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. ...

Hi, we have a fresh server with multi-tenant freeswitch and blue.box. We want to install on a separate box ASTPP and configure a set of options in it: - 3 wholesale trunks - LCR - Worls plans - DID management Also, in Blue Box, users would be able to access voicemail and modify any settings like forwarding number, voice mail and other typical things. Freelancer should provide 2 things after finish: 1.- A small manual to manage all this things 2.- A monthly fee to maintain and keep this system working. Thanks,

K zobrazení podrobností se zaregistrujte nebo přihlaste.

i have server with staticip , want someone install and configure freeswitch on my server, and also a billing solution for Freeswitch , future paid support is expected from that guy, VNC user and password details will be provided to do everything remotely , VOIP Gateway will also be provided task will be completed when a test outgoing call is to be made from freeswitch and billing is to be done

Need resource to add to current level of .net platrform working with Freeswitch functionality.

I need somone who can install freeswitch or any voip switch gatewap for secure VOIP calls using ZRTP So we can use IOS, android dialers for same thanks

We have provided automated scripts that installs Newfies-Dialer on your Ubuntu Server and includes all the dependencies such as Freeswitch, Celery and Redis -

We are looking a opensips and freeswitch expert for a specifically project. Do you have enough time to discuss project details ? Project is like that; timely opening ports on freeswitch shall send port available and time period (sec.) to open sips and ur billing-proxy server will send all calls to open sips. Opensips will manage all thess calls as available ports on frees with also freeswitch side need some arrangements for this plan. Will been shared after agreed on other parts.

We are looking a opensips and freeswitch expert for a specifically project. Do you have enough time to discuss project details ? Project is like that; timely opening ports on freeswitch shall send port available and time period (sec.) to open sips and ur billing-proxy server will send all calls to open sips. Opensips will manage all thess calls as available ports on frees with also freeswitch side need some arrangements for this plan. Will been shared after agreed on other parts. Values are entried only complete offer. If have any question please do not hesitate to ask.

We are looking a opensips and freeswitch expert for a specifically project. Do you have enough time to discuss project details ? Project is like that; timely opening ports on freeswitch shall send port available and time period (sec.) to open sips and ur billing-proxy server will send all calls to open sips. Opensips will manage all thess calls as available ports on frees with also freeswitch side need some arrangements for this plan. Will been shared after agreed on other parts. If have any questions please do not hesitate to contact me.

We are looking a opensips and freeswitch expert for a specifically project. Do you have enough time to discuss project details ? Project is like that; timely opening ports on freeswitch shall send port available and time period (sec.) to open sips and ur billing-proxy server will send all calls to open sips. Opensips will manage all thess calls as available ports on frees with also freeswitch side need some arrangements for this plan. Will been shared after agreed on other parts.

MUST FIRST DO OR I WILL NOT CONSIDER YOUR BID Sign up for demo account here - Look at ALL pages and functionality. I need the same thing. Pretty simple. Don't ask me dumb questions. Do your research on this site first. I already have Newfies installed. Just need added functionality and it customized with this Bootstrap theme - About Newfies - You must be an expert with the platforms this is built on. Thanks!

0 Frilansare som tittar just nu 5967 TOTALA ANTALET visningar Projektbeskrivning: We are currently looking a driven experienced PHP Developer for developing a flexible API in Laravel. First milestone in the project is to migration from Code Igniter and Asterisk (VoIP software) to Laravel and FreeSwitch. After that we will start at scalability. You will be a part of a small team where you together will drive our product in a nice fasion. You will together with out team look on how we can improve system designs, implement new features and debugging. We are always looking for new techknowledges to improve the user experienced so there will be time to test new things and validate them for production sites. Requirements: - You are fluent in PHP programming language ...

Hello, We are looking experienced programmer for developing users interface for freeswitch IP PBX. also need integrate it's with server installed freeswitch ipPBX Only iPPBX professional and expertise BID this project.

I have local mobile walet payment API's that i would like to integrate into an opensource calling platform, It is for a School Phone service and i need to integrate a local mobile payment solution API. I am deploying Freeswitch for switching and vbilling for billing. So it has to be integrated into vbilling as an optional payment solution. I am looking for someone versed in opensource technologies and knows freeswitch and vbilling very well. Rest based api knowledge is a plus.

Need sms integrated with my freeswitch/fusionpbx cloud environment - Looking for other enhancement suggestions as well for additional fee's

Recently We have installed Freeswitch with ASTPP billing system. At this time we need a sip softphone that should support above listed smartphones. Also we need configure VPN ( PPTP or L2TP . etc) inside the app in order to avoid blocking from ISP. If any one who worked with the subjected matter with excellent skills please contact me. SKILL Required ---------------------- Ability to develop sip softphone for listed smartphone. Ability to configure PPTP/L2TP or other VPN Protocol in CentOS 6.5 server (It should support more then 500 user connection to VPN ) Thank you

Please DO NOT bid unless you are familiar with each of the items described. We will only select someone that has Asterisk related feedback. Our server has Debian Squeeze with an asterisk application on it. We need some help on a few basics. Since I'd like to learn how to do these you must be open to showing me ...VOIP provider set in the system. I'd like to learn how to change this to another provider. - We also have mail forwarding setup on the server, and it currently works using gmail, but I'd like to use a different email provider and want to know how to change the settings. Configuration is done using: dpkg-reconfigure exim4-config. Keywords (please ignore these): Linux, Debian, opensips, freeswitch, voip, squeeze, ubuntu, network administratio...

Witam, Zlecę stworzenie systemu, który będzie pozwalał wykrywać ile telefonów komórkowych jest w danym (małym) zasięgu. Identyfikacja aparatów po numerze telefonu lub IMEI nie jest wymagana. Niepotrzebna jest też ich geograficzna lokalizacja. Wystarczy tylko ILE "KOMÓREK" JEST W ZASIĘGU. Bardzo możliwe, że można to wykonać za pomocą urządzenia USRP oraz oprogramowania OpenBTS. Mam je dostępne razem ze wszystkimi modułami potrzebnymi do komunikacji GSM. Nie musi być to jednak zrealizowane na USRP. Wszelkie pomysły dozwolone.

Recently We have installed Freeswitch with ASTPP billing system. At this time we need a sip softphone that should support above listed smartphones. Also we need configure VPN ( PPTP or L2TP . etc) inside the app in order to avoid blocking from ISP. If any one who worked with the subjected matter with excellent skills please contact me. SKILL Required ---------------------- Ability to develop sip softphone for listed smartphone. Ability to configure PPTP/L2TP or other VPN Protocol in CentOS 6.5 server (It should support more then 500 user connection to VPN ) Thank you

03:01.0 Ethernet controller: Digium, Inc. Wildcard TDM2400P 24-port analog card (rev 11) I need someone to install this fxo for outbound calls. So far issues we have had with these lines is very limited signaling everything comes as a message rather than tone eg invalid number number switched off number not...signaling everything comes as a message rather than tone eg invalid number number switched off number not available number is busy So when we use some fxo gateways on the market the billing starts on these messages. We need to ensure that billing only happens when there is a real call in progress. We had no luck with asterisk so far which is why I want freeswitch to be tried Please only bid if you have experience getting freeswitch to work wit...