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Sip Jobs

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Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
FreeSwitch integration with existing website and MySQL database using PHP. Purpose of integration is to develop contact management system that will send periodic voicemail messages to clients and manage follow-up messaging system. This development should include suggestions on how to improve existing MySQL database structure to facilitate on-going contact management. End user will be using the system to prospect for new clients by sending voicemail messages of pre-recorded audio messages. Once a voicemail message has been successfully sent, the PHP code must document the date and time that the message was sent to the prospective client so that a follow-up date is always updated for future voicemail broadcast. The end user should be able to choose the follow-up date for the entire list of recipients and, also, be able to remove recipients from the distribution list. The system should allow for multiple distribution lists at any time. The voicemail messaging system should be automated to the point that end user can put any client on a multi-touch voicemail broadcast campaign and the system will not send any voicemail message until a new audio message is uploaded by the end user. Please specify a lump sum amount for your proposal or you will be disqualified. Please let us know if you have any further questions.
Skills: SIP Asterisk CSS3 Freeswitch
Fixed-Price - Intermediate ($$) - Est. Budget: $300 - Posted
Hello, We are a VoIP Company and we are looking do design and create an VoIP Dialer using Open source. We want to hire someone who had experience and done work on creating app before. We are looking for a developer and designer. The person we are looking for most know that this is a long term job. We want them to maintain the App and do any update or upgrade in the future. We want to offer then a monthly payment.
Skills: SIP Android Android App Development Android SDK
Hourly - Intermediate ($$) - Est. Time: 3 to 6 months, 30+ hrs/week - Posted
I need some one who can install and manage this system in CentOS. I have a couple of servers that I need install and manage this system. You must be and expert in all aspects of Asterisk and Vicidial, FreePBX. I don't know which system I have to use. 1. Please suggest me which system should be used. 2. Please answer the following questions: - Advantage and disadvantages of suggested system. - Do you have detail document to install and configure of the system on CentOS? If you send it to me, then I will hire you immediately. if you don't have it, then you do not apply this job. - total time and your price. This job will proceed with on going for management of this system.
Skills: SIP A2Billing Asterisk CentOS
Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
We are looking for a highly experience FreePBX expert to configure our already well installed and secured VOIP FreePBX system. The task that need doing are as follows: 1. Configure Trunks 2. configure 8 incoming DIDs from 3 providers 3. Configure outbound and incoming routes 4. Setup 3 IVR, 5 extensions each, 3 Queues and call forwarding This is an urgent requirement that need to be completed ASAP. Only apply if you are free and can start straight away. You must also be familiar with the following DID and Voip careers: Voxbeam, Flowroute, Dinumbers.com. Happy bidding.
Skills: SIP Asterisk CentOS Elastix
Hourly - Expert ($$$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
On sites where we deploy Mikrotik routers we have discovered NAT issues for sip clients registering to freepbx hosted servers in remote data centres. ... It seems Mikrotik Nat is broken for sip. With routeros "sip helper" enabled or disabled, we have the problem of sip end points periodically showing "Un-Registered" If we replace Mikrotik with any other router like a snapgear or cheap dlink the problems go away.
Skills: SIP Asterisk Mikrotik RouterOS VOIP Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
We are seeking a Kamailio or OpenSIPS expert to help us install and configure a SIP proxy in front of multiple Asterisk servers. Currently, we have a cluster of 4 asterisk servers in a multi-tenant setup, but the end user needs to be told which one to connect to. ... The Asterisk servers connect to a MYSQL-based back-end (MariaDB Galera) The SIP proxy must: 1. Perform a simple MYSQL lookup based on the user ID to determine which Asterisk server this user belongs to 2. ... Pass through registration to the correct Asterisk server 3. Block mass SIP Registration Attacks 4. Provide NAT translation assistance 5.
Skills: SIP Asterisk MySQL Programming OpenSIPS
Fixed-Price - Intermediate ($$) - Est. Budget: $900 - Posted
For tunnel server it is expected the developer to use plain C language to write a flexible tunnel server . Its overall objective is to communicate with sip server to translate encrypted data passed from dialers. ... Android client api will be developed by java to integrate with android sip dialer. Iphone client api will be developed by Objective C to integrate with iphone sip dialer.
Skills: SIP Android App Development iPhone App Development VOIP Administration
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