Portaone openser työt

We are using PortaOne system. We need someone to develop certain functions and deploy them through web or desktop application, using their offered XML API. The functions are mainly reports that should be given to resellers and customers, through a web interface or sent by email on daily basis. There would also be some automated process that retrieves information about account daily at a certain time, and stores the information in a database or sends them by email (will be decided depending on the data size). We also need to put together a website for end-users, that shows them information about their account, and includes an integration with 2CO as an online payment gateway. Priority would be given to persons who already have experience with Portaone and 2CO APIs.

I have installed OpenSER and need everything verified to be working. We would like a front end manager, we are looking at CDRTools, asipto, or OpenSERadmin. We need you to recommend what to use and why. Ideally we want a front end interface that users can login and view CDR, rates, add endpoints, make payments etc. If you have or know of what to use we can go with that suggestion. Previous jobs in this area will have you considered first. Thank you for viewing and bidding :)

...thinking to use Openser to solve nat issues and also get a more scalable system. I will give you Teamviewer access. So i need the following: We need a serious professional to install and configure an OpenSER (Kamailio) integration with our Asterisk + A2Billing platform. - OpenSER 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 (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created in A2Billing. Also, each user should have their own voicemail...

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...sought to deliver turn-key service provider website and open-source server software. 1. Provide Admin interface - authentication, configuration, billing, service options, content management, reports, etc.. 2. Provide User interface - authentication, payments, customer self-service configuration, content management, reports, etc.. No need to reinvent the wheel. Asterisk, Freeswitch, Bayonne, OpenSer-Kamailio, OpenSips, SER, sipX, etc. have all been used for this type of application. Web development skills could be PHP, Ajax, MySQL, Jquery, LAMP, Zend, CRON, Javascript, HTML-CSS, Web 2.0, Cake, etc.. The application has these main components: 1. Obtain a DID number from a carrier API. 2. Forward this call if needed. 3. Simultaneous ring "find me" if needed. ...

I have two asterisk servers running openglue installed and running. Both are connected to a Carrier SIP Trunk receiving calls and running applications. I need someone to configure the Opensips ( OpenSER ) to load balance across the two asterisk servers, this trunks, some endpoints and two new Trunks from different carriers . When receiving a call to the opensips the opensips will route the call to the asterisk dependent on the load. This project involves configuring the Opensips only.

Dear Freelancer, We would like to build a new ...A2B, Mysql, web servers for the platform management. The architecture is designed to support the failover so all the configs are stored in the database. In total there is 8 servers to setup We have an old platform in other site with an old kamailio asterisks with 100 of users, we need to export sip info of all user to the new platform Only Freelancer with a strong experience with Kamailio (openser), linux admin & security, and asterisk are invited We will also need your service on long-term (monthly base) to monitor the platform and amelioration. We will send the schema of the architecture to you once you bid Our budget is 800-1000 The job should start as soon as possible and should be finish in 6 days T...

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We are looking for someone that is familiar with the Portaone softwitch to integrate the products to our website for online registration and purchases. ## Deliverables 1) All deliverables will be considered "work made for hire" under U.S. Copyright law. Employer will receive exclusive and complete copyrights to all work purchased. (No 3rd party components unless all copyright ramifications are explained AND AGREED TO by the employer on the site per the worker's Worker Legal Agreement). ## Platform Portaone softswitch / Linux / web integration

This is the problem i am having, load balancing is working fine. I need someone to fix the IP authentication kamailio in front of asterisk 1 and asterisk 2 load balance using dispatcher module. currently using redirect 300 302 but this requires additional setup on client gateway need to use t_relay or forward but be able to authenticate vi...of asterisk 1 and asterisk 2 load balance using dispatcher module. currently using redirect 300 302 but this requires additional setup on client gateway need to use t_relay or forward but be able to authenticate via source IP. Prefer custom sip header, but we are open to suggestion. need user/name authentication as well as IP authentication preferably on asterisk but it can be perform on openser without alteration of a2billing database ...

I need a VoIP PRo to install a SIP express router package with web interface on my Linksys ERT54GL router. This router is already flashed with OpenWRT and is ready to go. You may want to install openser, opensips or milkfish. All i need is a web interface from any of the above packages so I can register my SIP phones on it. SO the router will be used as a SIP server and then in turn with communicate with my asterisk box. I need you to: 1- install the package of your choice 2- Provide a web interface for the SIP server 3- Help me register one phone with the sip server 4- have the SIP server communicate wth my asterisk server ## Deliverables 1) All deliverables will be considered "work made for hire" under U.S. Copyright law. Employer will receive exclusive and complete c...

Dear All, As per this article http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms asterisk support PSTN to SMS and from SMS support endpoints to PSTN. In my scenario we have SS7 based E'1 on which our SMS providers are sending SMS on our DID numbers and my all DID's are registered on OPENSER. What I want to do I want to receive SMS from PSTN on E’1 and forward on Register user’s if Register Endpoint is SMS supported or Instant message supported then message should deliver on it or reject and vice versa if Endpoint generate SMS or instant message if should be forward on PSTN and related to own DID number’s message should not forward on PSTN if should forward internal. First I want to confirm it is possible or not, and second I want to do as project...

We need to integrate the Linkpoint gateway with our PortaOne mr19 switch. This equipment use the Business::OnlinePayment::LinkPoint framework .

Hello, Technical Details: We are looking for someone who has very good experience working with: 1) Openfire 2) Spark Client customization 3) Openfire Asterisk integration Preferably some Java webservices and OpenSIPS/OpenSER experience (These are not must!). Work Description: We have Openfire setup and integrated with our SIP Proxy and we need some help with the Spark plugins for VoIP. We have come across 2 plugins which we have tested - one using Openfire Asterisk and another Openfire and SIP. Please apply if you have proven experience working with Openfire and Spark Client Customization. Otherwise, please do not waste your time and our time. Thanks RM

Need an Opensips 1.6.x installation to be properly configured for 3 gateways with least cost routing. ? The gateways will require authentication. I will give you the gateways and login credentials as well as the rates. The Opensips server has been setup with the web control panel (opensips-cp) and is available via a public ip.? Work scope: -Set up and configure LCR module. -Change routing script to manage: ? ? -LCR routing ? ? -Gateway authentication ? ? -Failover in case original routing unreachable Thanks!

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Looking for a good systems administrator for Cisco, Mera, Asterisk, Openser to provide an on-going support for the VoIP Carrier. Job will include maintaining and setting up new VoIP carrier connections - SIP and H323. Work with carriers around the world. Configure new or Modify existing configurations, etc. Good knowledge of MySQL, Openser/Kamailo routing, knowledge Oracle is a plus. Work without supervision. A proven candidate must have 1 year experience, good GAF ratings and references.

We are locking freeradius billing system with admin, multilevel agent, customer & signup, calling card, call shop, wholesale module. Billing GUI must design in php or perl language & database in mysql. Actually, I m planning to start 1 server with freeswitch + freeradius billing& second with openser + freeradius billing. Developer has to provide documentation and source code for system.

Good systems administrator for Asterisk, Openser, FreeSwitch To provide support and installation help on a monthly contract basis. Configure new or Modify existing configurations, etc. Good knowledge of MySQL, Asterisk realtime, Openser/Kamailo routing, Work without supervision. A proven candidate may have 1 year worth of constant work. Good GAF rating, experience and record necessary.

On behalf of client in Scandinavia, we are looking for persons/small companies from the Baltics / Eastern Europe for long term cooperation. Fulltime, parttime and project positions. Wanted qualifications: - VoIP configuration and development (Asterisk, OpenSER/Kamailio) - iPhone development - Java development - LAMP development All should have good Linux and Networking skills. If you have one or more of these qualifications, please contact us. Meetings in Sweden/Norway is needed appr. once a month.

Customized elastix a2billing design or openser (kamailio)? The main idea is that several customers are going to use our asterisk based PBX. For each "slave" PBX all of internal users are going to have access to international phone calls and their PBX it is going to be link to our softswitch via IAX trunking due to DNS bug in Asterisk (if customer looses the internet connectivity all internal SIP phones are going to be dead one ore more SIP trunk not to be found , DNS resolution in chan_sip.c it is going to freeze up the system )? I need help in elaborating the specifications for a upcoming project. The coder is going to help me out with a network diagram , project specs , and the project will be reloaded to rentacoder for design/implementation? All the ideas are goi...

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Dear Sir, I'm searching for anyone that can install OpenSER + a billing platform on a dedicated CentOS server. The goal is to provide SIP termination from multiple originating hosts like betamax

Dear Sir, I rent a dedicated server that currently hosts Linux CentOS 5.4. I would like to launch a wholesale termination service and of course I'll have to buy credit from many SIP providers. I'm able to install OpenSER SIP server myself but I'll need a customizable billing+routing web interface that allows to: - Redirect calls using the cheapest route (so need of CSV import of each SIP provider) (LCR) - Automatically disconnect when user is out of credit (integrated with OpenSER) - Customizable Web interface with some API functions like signup/view CDR/refill etc... I'll pay for the remote installation, I'll have to customize the PHP website myself. PS: I'm not interested about Asterisk PBX based solutions.

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 the code.

...application (SIP UA) which I want them to register to OpenSIPS proxy server and have this proxy server to manage SIP trunks for outbound calling. I would like have capability to register and maintain multiple SIP trunks on proxy and use them based on predefined dialplan/rules. We must be able to setup and distinguish billing details details and CDR etc.. Please bid only if you have experience with OpenSER/OpenSIPs in past and confident to do the job. This is small bid and it can lead to long term relationship. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites...

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I have a deployment of multiple openser servers. I need to install and configure CDRTool to work with these servers to produce CDR records. 1. All post paid accounts 2. 2 OpenSer Servers and 1 asterisk server 3. just need to view, rate, and store cdr records in database.

I currently have an Openser (Kamillio) sip server running. When I receive an inbound call I am changing the alias_username from 7402492013 to username 8004172603 . Basically I am redirecting calls and changing the DNIS to what ever I want. My problem is that in sip message the To sip stays at original DNIS. See example below where i change DNIS from 7402492013 to 8004172603 and send invite out to next server. As you can see the "to sip" setting is still showing 7402492013. I want a change made to my config file that will also change the "to sip" setting to 8004172603. INVITE sip:8004172603@7.46.2.61 SIP/2.0. Record-Route: <sip:7.5.1.84;lr=on>. Via: SIP/2.0/UDP 7.5.179.82;branch=z9hG4bKe05a.7c19ecb7.0. Via: SIP/2.0/UDP 6.195.140.54:5060;branch=z9h...

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 the code.

Hi All, I need some help in a VOIP software portaone. If you dont have exp in VOIP it dosent matter, but you should be very good in PERL and MYSQL Please msg me for details. Thanks

Qualifications: - Kamailio 1.5.x/1.6.x - SIP Protocol - VoIP Experience (2+ Years) - Asterisk PBX Experience Description: We need custom configuration of Kamailio Server, with modules configuration as well, for small ITSP (300 Users), however it must be an efficient configuration to handle thousands of simultaneous transacti...* sqlops * tm * uri_db * userblacklist * usrloc * xlog * nathelper * nat_traversal This final configuration will be designed to interconnect to 2+ providers, and route SIP packets to carriers based on lcr, and other options defined. Full details we can go over after the project winner. This is good for someone that has worked at a ITSP or possibly a hobbyist/enthusiast of OpenSER/Kamailio with years of playing around... ...

Need expert consultancy and configuration to configure OpenSER + Asterisk + A2B (Approx 5 Servers) Must be able to configure to satisfy the following; [Billing Models Needed] Wholesale termination Peer to Peer (residential and business DIDs) Post and Prepaid billing Reseller Billing Calling Card Hosted PBX (an auto attendant hosted for a client) [Class 5 Features] Call Transfer Call Waiting Caller ID Call Forwarding Busy Call Forwarding 3 Way Calling Conference Return Call Follow-Me 911 Provisioning supported? [IVR Features] Auto Attendant (offered to clients) Voicemail Music on hold [Fax] Fax to email T.38 supported [DID] DID Pool real time - Customers use online signup to open account and select DID from pool Do you have a way to...

i am looking for extremely reliable resource available from 2PM GMT to 11 PM GMT with high level of expertise in Openser/Kamailio + knowledge of STUN, NAT, Proxy will be additional bonus. Thanks

I have kamillio installed on on of my servers that is up and running. It already have a database and authentication with trusted table installed but every call goes to one carrier. I would like to add least Cost Call routing to it. I would like to add multiple outbound carriers and add some type of priority to routes by destination prefix (LCR). I also need the cdrs to output to a table. I need this as soon as possible. Also one of my providers requires registration

We are looking for to outsource the management and surveillance of our telecommunications platform. Hosting of servers and broadband can also be indcluded but could also be taken care of by third party. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete sou...be considered "work made for hire" under U.S. Copyright law. Buyer will receive exclusive and complete copyrights to all work purchased. (No GPL, GNU, 3rd party components, etc. unless all copyright ramifications are explained AND AGREED TO by the buyer on the site per the coder's Seller Legal Agreement). ## Platform The management of the system requires good experience in Unix, Asterisk, OpenSER, SIP, PostgreSQL, MySQL, PRI ISDN and also preferably p...

...IP Telephony system a fully functional, standalone product. To achieve all the needs specified by our clients, including the deliver of a broad range of IP Centrex/Digital services and the need of core SIP integration with the Brazilian Telcos telephony infrastructure, this system will comprise a upper layer made by an Asterisk IP PBX and a lower, more infrastructural layer constituted by an OpenSER server. The attached diagrams 1 and 2 explain this concept better, on a visually accessible way. System needs and interfaces: Those are the system application units, system interfaces and connections as needed, according the prior shown diagrams: Core application units: Asterisk with custom contexts (multi-tenant) will provide digital services, device registration an...

Test and modify an existing Kamailio and Asterisk setup and install A2Billing. Verify operations of H323 in the Asterisk setup and Kamilio routing (Round robin and by DNID). Round robin already done.

We are looking for somebody who can integrate Asterisk, A2Billing 1.4, and Kamailio together (real time DB for all). We expect A2Billing to be the controller/billing for Kamailio and Asterisk. We also expect a test period of your services to make sure things work as expected. If you have installed this configuration successfully (used in a retail/production environment) please consider bidding. It would also be helpful if you are good customizing the above for future expansion of our services as noted below.] Note: These are not included in current bid unless you indicate so. Call Shop Click2Call Web C2C button that charges the button owner a click fee plus call time. The owners ID and or PW cannot be visible to any browsers. Reseller (in addition to agent that A2billing 1...

I am looking for somebody who can integare Asterisk A2Billing with the OpenSER. If you have done this before please contact. here is what we are trying to do. OpenSER (Proxy WebServer (A2Billing Web) MySQL )A2billing DB) Asterisk 1 Gateway Asterisk 2 Gateway You need to install Asterisk, A2bIlling, OpenSER and need to make a calling card platform work

Systems Administrator for VoIP platform including Asterisk, OpenSer, Freeswitch. Test with other carriers working as a team verifying call routing. Develop and clean up dial plans, test existing setup etc. Test existing Asterisk/OpenSer configuration, Create or Modify Asterisk dialplan and OpenSer configuration.

I am re-posting this project due to failure from provider. We need a serious professional to install and configure an OpenSER (Kamailio) + Asterisk + A2Billing platform. - OpenSER 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 international calls (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created/added in A2Billing. Also, each user should have their own voicemail account and OpenSER should route unanswered/failed calls to the user's voicemail. - Wholesale inbound calls should b...

Good Day, We are looking for an Asterisk/A2Billing/OpenSER administrator to work on an hourly basis to review our VOIP plan, make recommendations and be available to make future modifications. We have an Asterisk Cluster using OpenSER as a sip router and a2billing as a billing engine and Asterisk for call handling. Our cluster is currently running. We need an experienced Asterisk/A2billing/OpenSER administrator who can make appropriate changes to our dial plan, create sip trunks to/from SIP Providers. We have a complete VOIP cluster plan that needs to be reviewed and revised so that we may properly structure our product offerings to match our services. If you are interest in this project please provide your hourly rate and skill set so that we may evaluat...

Good Day, We have an Asterisk Cluster using OpenSER as a sip router and a2billing as a billing engine. Server 1 - OpenSER Server 2 - MySQL Server 3 - Asterisk (there are actually 4 of them with 3 off until we get this routing problem fixed.) Server 4 - A2Billing We were configuring for incoming DID and somehow broke this system. AGI used to route all calls out, if credit was available. AGI now always prompts for language and asks the user to input a pin. We need our asterisk cluster to accept a sip registration, check against mysql/a2billing and route according to balance. Bal > 0.0 should route the call in or out. Can you assist? Inquire within. Thanks in advance, Thomas

Hi All, I'm looking to setup OPENSER service with AAA radius, i need openser configured with Accounting and Authentication via our billing software. Then routed to Our Asterisk or other gateway we have. Following features : Call routing to asterisk Voice mail to asterisk AAA radius via our billing system Thanks, O

We carry minutes between mobile operators and wholesale voice companies. We have so far done development of our platform in house. We are looking for someone to design / redesign several components: - An OpenSer/Sips front-end that functions much like a border gateway controller to back-end Asterisk. Some calls are SIP-to-SIP, some are SIP to IAX2 (or IAX2 to SIP (or even IAX2 to IAX2)). Scalable multi-servers, load balancing, HA required. Must have LCR component based on custom MyRates system. - A rating engine based on custom PHP software and Mysql Run every 5 minutes to rate calls based on asterisk cdr records Redesign DB design for HA, failover etc. Billing based on different interval, PDF invoice generation, supplier billing reconciliation. - A Web P...

We need a serious professional to install and configure an OpenSER (Kamailio) + Asterisk + A2Billing platform. - OpenSER 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 international calls (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created in A2Billing. Also, each user should have their own voicemail account and OpenSER should route unanswered/failed calls to the user's voicemail. - Wholesale inbound calls should be authenticated by IP address.

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We would like to have developed a browser plugin that processes webpages and finds telephone numbers and marks them so the person browsing the webpage can click on the numbers to start dialing them through our PBX. The browser plugin should be for Mozilla Firefox 2.x and 3.x and Internet explorer. The dialing should be done through a call API which can be accessed on our PBX ( asterisk + openSER ). ## Deliverables For an example of this module you can find it at : We use the same PBX system as well. There are 2 important things : 1. The 4psa browser module makes boxes of telephone numbers. We also want to be able to select a number with your mouse as text and then pressing the right mouse