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Asterisk Jobs

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Hourly - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
1) Build a system using YATE and/or Asterisk that allows MGCP-based ATAs at customer locations to utilize SIP trunks. 2) Make it feasible to offer voicemail 3) OPTIONAL: create the concept of a calling queue or ring group that can include both SIP extensions and cell #s, where the Cell #s can hear caller ID

Fixed-Price - Est. Budget: $ 150 Posted
Dear freelancer, We buildup a pbx system by asterisk on centos 6.6, and setup some extensions for they can call each other. We need to know realtime status(online/offline, on-calling/available) of each extensions, and it syncs the realtime status to mysql specific table, then we can use it for applications. We need the solution detailed procedures for next time we can rebuild our pbx system by ourselves. Thanks in advance. Best Regards,

Fixed-Price - Est. Budget: $ 90 Posted
Interactive Voice Response (IVR) VoIP for Inbound. 1. Re-configuration of the current Asterisk. (Due to migration from old to new server) 2. clone existing system to create new one 3. The newly cloned system to change name. and add extra line of script to the dialplan NOTE: - Both data (customer name, account number, pin and balance) must connect to the tomcat mysql database. - Configuration of G.722, G.729 and also TTS (Text to Speech) engine.

Hourly - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
The TCP/IP VOIP Engineer will be providing functional and practical analysis related to the installation of the communication networks. The TCP/IP VOIP Engineer will be given the responsibility of Designing, maintaining and improving voice and telephony infrastructures and ensures that all are secure, available (99+% minimum uptime) and functioning efficiently. You will be able to ultilize your knowledge of commonly used communication concepts, practices and procedures to diagnose and resolve moderately complex communication issues. Coordinates strategies for defining, deploying and maintaining the companies voice and video communication architecture and its associated network connections and component hardware. The TCP/IP VOIP Engineer also manages all engineering projects for voice and video initiatives, planning technology roadmaps and configuring and optimizing all PBx telephone system and services. Additional skills needed: *excellent customer service

Fixed-Price - Est. Budget: $ 10 Posted
Hello. Today I'm looking for a professional Asterisk/FreePBX programmer, that knows about DialPlan writing and maybe other interfaces of Asterisk. I need a simple application that allows users to log-in the voice menu, and be able to record a message, or to scroll (using the touch-tone phones) through messages others were left for the community. It's like a primitive BBS or Internet forum, but for use by people over the phone. It seems to me it can be achieved by dialplan programming. but if an external application server is better - I think it will be good for me also.

Fixed-Price - Est. Budget: $ 2,000 Posted
The objective of the project is to modify the existing FaxServer solution based on Hylafax and Asterisk. Current solution supports T.30 protocol with SIP Trunks and G.711 (Asterisk with IAX modems). Modified solution should provide SIP interface (SIP Trunks) with T.38 (we require the usage of Freeswitch - www.freeswitch.org). Modifications should be transparent for other system functionalities. The software to be used should be released under Open-Source License (excluding GPL,CPL).

Hourly - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Our company currently uses over 1,000 different VOIP local phone DID numbers. We currently pay .99 per number per month through Ringcentral. Our plan gives us the minutes we need (around 15K/month) and 5 users, a phone tree, departmental extensions, personal extensions, automatic voice recording, and softphones for around $1,500 per month. We are looking for a VOIP consultant that can get us a better way at a lower cost. In addition, We hope this person will also help us keep our 1,000 current DID numbers. Again we are looking for a way to pay less than .99 cents per number for our 1,000 local number or a way to pay one up front fee for permanent use of our current numbers. I assume that we will need to port these to the new system. We don't know if we need another VOIP provider, and SIP trunk provider, just a DID host, our own ISP license, Asterisk or WHAT?? We are not sure if there is a way to do this but are interested to find and will need help with the process. Thanks...

Hourly - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We recently subscribed to a hosted Asterix server with the PIAF (PBX in a Flash) User Interface. We need your help to configure it for routing inbound calls and teach us how to manage it going forward. The server is hosted at Vitelity. If you have prior experience with Vitelity's hosted servers, that would be awesome. Thank you. AK