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Asterisk Jobs

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Hourly - Expert ($$$) - Est. Time: Less than 1 month, 30+ hrs/week - Posted
We are looking for a Freeswitch developer to help replicating our existing Asterisk Custom Proprietary API features which is coded in PHP and recreate these features in Freswitch API code it in RoR Objective is to replicate the existing feature and then develop the new features on Freeswitch API platform such as Postpaid Billing, Agents module, etc. ... After this new platform is created we will stop using our existing Asterisk system. The qualified candidate must have extensive experience in Freeswitch and Asterisk and understanding of FreePBX, MYSQL and PHP and some RubyOnRails. ... The qualified candidate must have extensive experience in Freeswitch and Asterisk and understanding of FreePBX, MYSQL and PHP and some RubyOnRails.
Skills: Asterisk FreePBX Freeswitch
Fixed-Price - Intermediate ($$) - 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: Asterisk Android Android App Development Android SDK
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
FreePBX/Asterisk phone system (Linux-based, Open Source). Need to create a script that will export the Directory data (extensions, names) to then create a file that the Polycom phones would download upon boot (via TFTP or FTP).
Skills: Asterisk FreePBX
Fixed-Price - Expert ($$$) - Est. Budget: $200 - Posted
I need to create a new Elastix PBX. Now I have one in my private server and i want to change to Digital Ocean, linux server. I use tree number, 1 for Mexico, 1 for Colombia and 1 for Perú. The number works with redvoiss. http://www.redvoiss.com/ I have a queue for each country (México, Colombia y Perú) and queue for each annexes. also, I need advise, with how can we protege the PBX from hacking.
Skills: Asterisk Elastix Linux System Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
We are looking for a highly experience FreePBX expert to configure our already well installed and secured VOIP FreePBX system. The task that need doing are as follows: 1. Configure Trunks 2. configure 8 incoming DIDs from 3 providers 3. Configure outbound and incoming routes 4. Setup 3 IVR, 5 extensions each, 3 Queues and call forwarding This is an urgent requirement that need to be completed ASAP. Only apply if you are free and can start straight away. You must also be familiar with the following DID and Voip careers: Voxbeam, Flowroute, Dinumbers.com. Happy bidding.
Skills: Asterisk CentOS Elastix FreePBX
Hourly - Expert ($$$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We have recently moved to Mikrotik for our routers. We have many VoIP clients. 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. Here is a link to a person with a similar problem: http://forum.mikrotik.com/viewtopic.php?t=79039 We would like to consult with you if you think you could help us solve this issue as we would like to use Mikrotik routers because they are well priced and full of very helpful features.
Skills: Asterisk Mikrotik RouterOS SIP 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. ... 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.
Skills: Asterisk MySQL Programming OpenSIPS PHP
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