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Sip Jobs

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Hourly - Entry Level ($) - Est. Time: 1 to 3 months, 10-30 hrs/week - Posted
What i'm interesting in is developing a custom audio gateway plugin that would allow audio to flow in and out to SIP enabled topologies (like asterisk/Etherstack and many others) . ... The AUDIO IN/OUT and Push-To-Talk and Incoming Voice Call is exposed in the API layer, which may allow use to code to an open SIP stack. I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). ... I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). Note the current development will be in windows, but possibly moved into windows embedded later.
Skills: SIP Asterisk Microsoft Visual Studio OpenSIPS
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system ... We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system
Skills: SIP Asterisk CentOS FreePBX
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
Customize vicidial according to our requirement for inbound callcenter. Customization includes 1. Look and feel of the agent and admin/supervisor interface. 2. Simplified method to login to the system 3. Simplified screen to create/manage agents and extensions.
  • Number of freelancers needed: 3
Skills: SIP Asterisk FreePBX JavaScript
Fixed-Price - Expert ($$$) - Est. Budget: $250 - Posted
Fax Exchange System: Configured Hylafax, IAXmodem, and Postfix for sending and receiving fax in ASTERISK (FreePBX/Elastix). This is Asterisk based fax exchange system which receives and send faxes. This system receives fax in TIFF format and then it converts it to PDF and emails the received fax to a specified email address. The frontend of this software need to be designed in c# and all these configuration were done in Elastix. This program make calls from c# which uses Asterisk at backend . It also uses MYSQL Server from which it get phone number and then it pass this phone number as variable to the Asterisk Dialplan, then a call is generated on that number using Asterisk. And then this software updates call status in the database (MySql server). Need a campaign monitor of campaign a user friendly.
Skills: SIP Asterisk C# Elastix
Fixed-Price - Intermediate ($$) - Est. Budget: $150 - Posted
Installation and configuration Asterisk v13 FreePBX v13 and A2Billing v2.2 on a remote vps or Installation and configuration freeswitch fusionpbx with billing installation on the vps if you're available.
Skills: SIP A2Billing Asterisk CentOS
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
We are looking for a Linux Server Administrator that specialises in both VOIP and VPN services. The person must have knowledge and experience with Asterisk VOIP and OpenVPN services. The job will be on a per needed bases. The administrator must have excellent English communication skills and be able to respond to technical support tickets from time to time.
Skills: SIP Asterisk CentOS Debian OS
Fixed-Price - Entry Level ($) - Est. Budget: $500 - Posted
Hi all, we are looking to work with someone to develop our own and unique telephone conferencing system. We have our own DID numbers and are looking to make use of these by creating our own telephone conferencing platform. Our platform needs to be unique and as automated as possible with a very simple to use customer / client interface. We require the person to develop the front end (website) and the backend (telephony server). Our typical client will access / request use of the service through a desktop website which also must be viewable on tablets and mobile devices. If you message me I can provide details of a comparative website / service to give you a good idea of exactly what we are looking for. We want our service to be more intuitive, offer more features and be more flexibility than the comparative website / service. The developer needs to be very committed to this project offer flexibility and be proud of the work they do. The right person will also be instructed on future works / projects. It should be noted I am available for contact 24/7 however I am best available UK time 12:30 to 13:30 and after 19:00 hours daily Monday to Friday and after 17:00 at the weekends. The developer can be located anywhere, however must be able to converse well in English. The price quotes is only a reference price I await for your bid and proposal.
Skills: SIP English Telephone Handling VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $800 - Posted
We want to build a mobile application, intending to provide a way of communication between worldwide consultants in various fields and those who’re seeking for a paid consultation in a certain field (Legal, medical, technical, ...etc.)... This will be done by providing VOIP connection between the 2 parties after agreeing to some terms and paying fees Also scope includes building a simple website for back office functions Please read details in attached document carefully And please note that we're negotiating the scope for the MVP only (strikethrough lines are not included in scope)
Skills: SIP Asterisk VOIP Software
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
Using the tool SIPP, I need to have a list of phone numbers in CSV and/or TXT format. It will need to execute standard UAC scenario against each number, adhering to a concurrent call limit and produce results for the list, including times. The intent is to: 1) validate that each number properly connects and provide results on a per number basis as rows in a output. 2) that the destination host can handle X simultaneous calls I believe this is all already possible with variable injection, however I would like someone to just provide a complete example of how to achieve this. Additional SIPP work and SIPSAK work will be needed over the next few weeks. Please have skills for both of these programs.
Skills: SIP Linux System Administration
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