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Sip Jobs

33 were found based on your criteria

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Hourly - Est. Time: More than 6 months, 30+ hrs/week - Posted
We are using Bicom PBXware MT system and we are looking for someone to help our customers with tech support on live chat and phone, the system is very easy to learn and perfect English is required along with some experience with Asterisk/Hosted PBX

Hourly - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
I am looking for someone to set up Bicom single tenant, We have many installs and would like for you to get a build sheet form a sales person and then build the PBX to the customer’s needs. Then also help with the remote install if needed. Knowledge with Pfsense is a plus. Networking and firewall

Fixed-Price - Est. Budget: $ 2,000 Posted
We are looking for skilled people who have proven experience with developing SIP VoIP Apps for both Android and iOS. For example an APP like Viber Only apply if you have developed one of these type of APPs before. The knowledge must be more than just UX. You need to have experience with integrating a SIP Stack with both iOS and Android so that the resulting APP integrates well with the the services and features of that operating system. The initial part of this process will be find candidates who can demonstrate their knowledge in developing such applications. To apply please: 1) List the SIP VoIP APP(s) you have developed along with a reference we can contact to confirm you wrote the APP. 2) Explain which SIP Stacks you have used and integrated in both iOS and Android APPs 3) Describe in detail the level of integration you managed to achieve between the SIP VoIP APP and the operating system and its services (iOS and Android). 4) List the SIP servers you have conducted...

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

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...

Fixed-Price - Est. Budget: $ 2,500 Posted
Essentially what we're looking for is an app just like Zoiper, but much more bare-bones and with our own company logo instead. Zoiper: http://www.zoiper.com/en/screenshots#mobile-ios http://www.zoiper.com/en/screenshots#mobile-android Main idea is that it should be an app that can be downloaded to a smartphone through the AppStore, Google Play store. User then launches the app, goes to Settings and creates a new SIP account by entering all the information we will provide to them manually such as: Host (IP or domain name of our VOIP server) Username and Password (username and password the user will try to authenticate with against our server) And maybe some other optional settings to make sure calls get routed through. Then they hit big Register button and get authenticated by our own VOIP server we already have in place running Asterisk. After they get authenticated by the server with the given credentials they get sent to a normal dial-pad/number-pad view where they...