Freeswitch Jobs

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Fixed-Price - Expert ($$$) - Est. Budget: $600 - Posted
We are looking for an experienced engineer/developer with knowledge in Siemens OpenScape/HiPath and VoIP. The work is to configure the Siemens OpenScape PBX (provided by freelancer, this can be a demo o temporary licence for the poc only) in a lab environment. Configure 3 extensions/softphones (3 SIP). no need to configure for external calls. Once this is up an running, we will need the freelancer to setup Orecx or similar packet sniffer to create voice and metadata files for the recording calls. We will provide port span configuration. Integration with CSTA will be a plus but must be optional. Then using Web Services (API will be provided) you will need to create a service running in the background that will send voice and metadata to a remote machine. Please do not apply if you have no experience on this. To apply start the covert with "I am a dreamer". You will need to provide the documentation and the code needed to accomplish the task.
Skills: Freeswitch Call Handling Network Engineering Telecommunications Engineering
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
I would like somebody to help me with the initial setup of a Freeswitch server. I want to use this as a conference call server (using mod_conference). https://freeswitch.org/confluence/display/FREESWITCH/mod_conference I want: * To be able to dial in using POTS to a conference call on any of the 3 DID numbers I provide which can be configured at DIDLogic ... I want to use this as a conference call server (using mod_conference). https://freeswitch.org/confluence/display/FREESWITCH/mod_conference I want: * To be able to dial in using POTS to a conference call on any of the 3 DID numbers I provide which can be configured at DIDLogic ... There doesn't need to be a concept of leader / leader PIN or database connection or anything complex at this stage, I am just trying to get the basic setup right. * To be able to 'originate' a call from the freeswitch CLI (fs_cli) using the 'originate' command which should dial out to a specified POTS number and patch them into a specified mod_conference room
Skills: Freeswitch Asterisk Linux System Administration SIP
Fixed-Price - Expert ($$$) - Est. Budget: $800 - Posted
Once this is up an running, we will need the freelancer to configure the phones for IPDRLINK. If the freelancer has knowledge on Freeswitch we would like to record the calls using Freeswicth . ... We can provide virtual machines in the cloud to install the Alcatel PBX and Freeswitch software if necessary. The job that needs to be done here is the installation and configuration of Alcatel OXE to work for one softphone calling the other. ... Setup recording for the softphones. Install and configure Freeswitch, and connect for recording with the Alcatel softphones.
Skills: Freeswitch Network Engineering Telecommunications Engineering VOIP Administration
Fixed Price Budget - Intermediate ($$) - $200 to $300 - Posted
Plivo integration with existing website and MySQL database using PHP. Purpose of integration is to develop outbound voicemail system that will send periodic voicemail messages to clients and manage follow-up messaging system. This development should include suggestions on how to improve existing MySQL database structure to facilitate on-going contact management. End user will be using the system to prospect for new clients by sending voicemail messages of pre-recorded audio messages. Once a voicemail message has been successfully sent, the PHP code must document the date and time that the message was sent to the prospective client so that a follow-up date is always updated for future voicemail broadcast. The end user should be able to choose the follow-up date for the entire list of recipients and, also, be able to remove recipients from the distribution list. The system should allow for multiple distribution lists at any time. The voicemail messaging system should be automated to the point that end user can put any client on a multi-touch voicemail broadcast campaign and the system will not send any voicemail message until a new audio message is uploaded by the end user. Please specify a lump sum amount for your proposal or you will be disqualified. Please let us know if you have any further questions.
Skills: Freeswitch Asterisk CSS3 HTML5