Tinyerp openserproiecte

Desejo realizar a integração via API da ferramenta de emissão de boleto Asaas com o TinyERP para eu usar meu banco de dados dos clientes do TinyERP para gerar automaticamente os boletos de cobrança pelo Asaas. Sites para suporte:

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 ...

Preciso integrar o estoque de 3 sistemas usando suas APIs. A integração: a. TinyERP -> eComanda: usuário entra com # da venda do Tiny e API carrega produtos no eComanda b. eComanda -> NuvemShop: sincronizar estoques a cada X horas c. NuvemShop -> eComanda: evento de venda no NuvemShop diminui as unidades no eComanda. Desenvolvimento deve incluir todos os alertas de erro possíves, exemplos: produto inexistente, # de venda já lançado, API indisponível, etc.. Procuro solução turn-key: Desenvolver código, implantar integração na nuvem e me entregar tudo funcionando. Só quero os números de estoque. Não preciso de im/exportação de produtos. Trabalho de, no m&aacu...

...(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...

...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...

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 ...

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 ...

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. ...

...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...

TITAN - TinyERP We are a small company in Brazil that are building a small ERP so we can improve our processes in order to have a better understanding of how we work and what we need to do to grow. We work in the construction industry, selling services and goods to contractors. This will be a web-based system with AngularJS + PHP (Codeigniter) + MySQL. We need modules like: - Customer - Supplier - Financials - Employee - Contact management .. and more.. = ALL THE DETAILS ARE IN THE ATTACHED FILE =

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...

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

...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...

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...

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

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

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

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

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

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

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

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

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

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

hello 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

we need to install kamailio + xmpp module + asterisk + a2billing The summary: 1. Audio Call 2. Video Call 3. Chat/ File Transfer 4. IVR, Voicemail 5. Call Plan, Billing, Voucher ,top Users will create an account and be able to call audio + video and chat with themselves and other the network (PSTN) We need to work with an experienced developer only .

Need someone to: 1. setup Kamailio (openser) sip server. 2. Configure to disable authentication for all requests 3. an API to sign-up new sip account

We currently operate OpenSER 1.2 as a Media Proxy and SIP Registration server back-ended to Asterisk 1.4. We have PERL scripts operate a HostedPBX platform we have written ourselves within Asterisk and a web interface based on PHP and MySQL. We require assistance to set up a new server which will need to operate on the EC2 platform of AWS Sydney. We would like to use CentOS 6.4 as the base operating system. We would like to use SIP-Router project to replace OpenSER 1.2 and also upgrade to Asterisk 11.x LTS as it is supported for another few years (later to upgrade to 13.x probably about 9 months after initial release to ensure stability). We need to ensure the following Asterisk functions work through the SIP-Router Proxy: * BLF Lights for calls made, transferred and i...

VICIdial-Asterisk-Se...server and SIP registrant, redirect server. • Specialized in collecting information from User/Client/Management by asking questionnaires to understand their requirement and providing the appropriate solution and for VOIP technology. • Having Good Knowledge on Linux Commands and Administration. • Good Understanding of VOIP, PRI, E1, T1, SIP, IAX protocols and implementation. • Good Understand IN OPENSER, KAMAILIO Like SIP based Registrant server. • Successful Implemented VICIDIAL, ELASTIX, A2BILLING like system as consultancy work. • More Than 15 Successful IVRS like system or PBX Implementation with 3 or more clients as consultancy work. • Expertise in ZAPTEL, DAHDI, Asterisk Implementation and Integration To any call center ...

VICIdial-Asterisk-Se...server and SIP registrant, redirect server. • Specialized in collecting information from User/Client/Management by asking questionnaires to understand their requirement and providing the appropriate solution and for VOIP technology. • Having Good Knowledge on Linux Commands and Administration. • Good Understanding of VOIP, PRI, E1, T1, SIP, IAX protocols and implementation. • Good Understand IN OPENSER, KAMAILIO Like SIP based Registrant server. • Successful Implemented VICIDIAL, ELASTIX, A2BILLING like system as consultancy work. • More Than 15 Successful IVRS like system or PBX Implementation with 3 or more clients as consultancy work. • Expertise in ZAPTEL, DAHDI, Asterisk Implementation and Integration To any call center ...

Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729,...Description: 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 ...

Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729,...Description: 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 ...

...Calling Card/Call back Solutions. SPECIFIC RESPONSIBILITIES: * Handling a Team of 20 people along with Team Leaders, Project Leaders. * Designing and implementing new features in architecture. * Bug fixing and enhancements * Linux Kernel Tweaking for kernel optimization * Designing and implementing MySQL Cluster database architecture with backup and storage policies. * Optimizing OpenSER and RADIUS for registering and authenticating multi-million users. * Testing and Optimizing different entity in architecture * Database Administration of MySQL, Optimizing and Tuning different parameters of MySQL server variables. Also performing Replication for Master / Slave architecture. * High Availability (Linux HA) of each component through heartbeat and scripting * DNS Loa...

I'm a VoIP engineer ,My company provides support to customers that want to implement opensource voip platforms in their environment.I support Freepbx,Asterisk,A2billing,Netfies,PBX in a flash ,Openser ,Freeswitch,Elastics..Basically any opensource platform.. I also offer Wholesale VoIP to customers that use my platform ,My idea is to capture the business user that wants to manage his own voip system and i provide support services to them..Take a look at Dr.VoIP.. I want to use the same persona...

I am looking for experienced person that have knowledge on Kamailio OpenSER (SIP Proxy) and can provide online training via Skype for few sessions. You must have good english to communicate and explain well via slides or any material.

I am looking for experienced person that have knowledge on Kamailio OpenSER (SIP Proxy) and can provide online training via Skype for few sessions. You must have good english to communicate and explain well via slides or any material.

Seeking assistance in setting up a Kamailio server to act as a outbound sip proxy.. Basically we want the signaling for outbound calls from several servers to pass through this proxy so we can give our carriers 1 IP address. Also our main Lampa dev will need to be trained in the basics of how you setup the kamailio config. Setup in Centos 6.3 Calls originating from Asterisk 1.8 1 carrier to start (simple forwarding) No RTP proxy

Calling Card and piness Solutions platform I am looking professional to develop the e commerce website, fully functional Online Calling card SIP-based Platform for 120 concurrent calls option to increase in future( Asterisk, SER, OpenSer, FreeSwitch) ""customers can create a calling account on website and be able to recharge it at any time, instead of having to repeatedly purchase a new calling card. They will have a local access number to dial to make long distance calls, but will never have to enter a PIN because the system will recognize them based on the number they are calling from, using automatic number recognition. They can sign up all the numbers they will use to call – their home, cell, and office numbers – and recharge their accounts at any ...

I want someone to train me for Vmware asterisk and openser to start a PBX hosting in US Trainer should know the concept of all voip related thing. My Aim is to start a hosting service in US and I am experienced in Asterisk support Only serious bids has to come Thanks

We have an urgent project in hand. Candidate should know Asterisk, OpenSER, Communigate, XMPP and VOIP. Interested can reach me on prabhat The Administrator removed this message due to contact details which breaches our Terms and Conditions.

We currently have a single A2billing server with a few clients on it. We are expecting the numbers to grow soon. We are interested in knowing more about creating a multi-server cluster using OpenSips/OpenSer to redirect the calls like a routeur to different server in fonction off the amount off calls on the servers or the cli off the calls or hours the time off day to different sip server like Asterisk/Freeswitch. The cluster should have a single instance of A2billing so that we can manage everything on one interface but the calls should be load balanced over multiple servers. We are currently capable of doing over 400 max simultaneous calls with long duration calls on a single A2billing server. We need to be able to load balance the calls from the OpenSip server to d...

Seeking assistance in setting up a Kamailio server to act as a outbound sip proxy / session boarder controller.. Basically we want the signaling for outbound calls from several servers to pass through this proxy so we can give our carriers 1 IP address. We dont want this proxy to be in the udp media stream.

We are seeking a Kamailio experienced system admin or dev that can help is setup a single Kamailio (openser) sip server in our dev environment. Functionality needed from kamailio setup is LCR, CDRs, and failover between carriers. Please only kamailio experienced freelancers only.

Goal of project is to redevelop ASTPP Voip Billing GUI in PHP MVC Framwork! 1)Redevelop ASTPP, Perl and CGI GUI from scratch in PHP 2)All ASTPP existing functions must be 100% ported to PHP 3)ASTPP supports Asterisk, Freeswitch and openser, please keep this in mind when developing the GUI.(As new GUI should support all functional softswitches) 4)ASTPP supports third party applications example: FreePBX, Thirdlane, Fusionpbx and osCommerce 5)We would like to develop support for: Bluebox and Whistle from Please note GUI should be develop in a modular design to support third-party plugins. It must use some sort of PHP framework to develop GUI. We will provide Administrator template which will need to be customised using mvc framework, jquery and ajax. Template that we

I'm looking for someone experienced in Kamailio to help me with my Kamailio 1.5.x based LB configuration ( module). I've got a SIP server and 2 media GWs; load balancing of incoming invites is working ok as long as there is no RTP, but when audio is established it all gets messed up. I think solution should be quite simple for a Kamailio/OpenSER expert.

Calling Card and piness Solutions platform I am looking professional to develop the e commerce website, fully functional Online Calling card SIP-based Platform for 120 concurrent calls option to increase in future( Asterisk, SER, OpenSer, FreeSwitch) ""customers can create a calling account on website and be able to recharge it at any time, instead of having to repeatedly purchase a new calling card. They will have a local access number to dial to make long distance calls,  but will never have to enter a PIN because the system will recognize them based on the number they are calling from, using automatic number recognition. They can sign up all the numbers they will use to call – their home, cell, and office numbers – and recharge their accounts at...

I have several Asterisk/A2billing servers with central database. I need to set up a transparent proxy (load balancer) using Kamailio/OpenSER so that calls are routed to available servers in round-robin fashion. I don't need registration on the proxy. I would also need training on how to manage the server and how I can add/remove Asterisk servers. You need to have good English speaking skills and experience with similar projects.

Goal of project is to redevelop ASTPP Voip Billing GUI in PHP MVC Framwork! 1)Redevelop ASTPP, Perl and CGI GUI from scratch in PHP 2)All ASTPP existing functions must be 100% ported to PHP 3)ASTPP supports Asterisk, Freeswitch and openser, please keep this in mind when developing the GUI.(As new GUI should support all functional softswitches) 4)ASTPP supports third party applications example: FreePBX, Thirdlane, Fusionpbx and osCommerce 5)We would like to develop support for: Bluebox and Whistle from Please note GUI should be develop in a modular design to support third-party plugins. It must use some sort of PHP framework to develop GUI. We will provide Administrator template which will need to be customised using mvc framework, jquery and ajax. Template that we

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. ...

I have an a2billing + asterisk 1.8 Running on Amazon EC2 But i have issues with nat clients so i'm thinking to use Kamailio to solve nat issues and also get a more scalable system. I can give you Teamviewer access. I need the following: We need a serious professional to install and configure Kamailio integration with our Asterisk + A2Billing platform. - Kamailio should act as Registrar with Nathelper + RTPProxy/MediaProxy and it should allow local users to call each other without passing through Asterisk. It should also route local, national and international calls like to Asterisk + A2Billing so i can still bill them. - Kamailio should authenticate users using the Asterisk/A2Billing database - Asterisk + A2Billing should be configured so that new user accounts wou...