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Fixed Price: Not Sure   |  Posted: May 23, 2015  |  Ends: 13d, 18h  |   2 Proposals
Need A simple hello world application with PJSIP stack. 1. PJSIP Audio Only Build 2. G729 Codec Support 3. STUN Support 4. VPN Support {T_FP_QUESTION0_START} Job description: OS version for Android: Android OS 4.0 (Ice Cream Sandwich) Development of webservices: Yes Development of graphic design and UI elements: I will provide the all assets.
Category: Mobile Applications       
Skills: Android       

e****ech
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| e****ech
|    Bangladesh
Hourly Rate: Not Sure   |  Duration: Not Sure  |  Posted: May 23, 2015  |  Ends: 13d, 17h  |   3 Proposals
Looking for an experienced consultant with deep technical skills in VoIP / Call Center. Must have deep experience on one of the following platforms: Cisco, Avaya, Genesys, Microsoft Lync. Or other VOIP platform Must have experience with TDM<->SIP voice gateways such as audiocodes and hardphones / TFTP. Working knowledge of queues, skills, call routing, IVRs Knowledge of SIP and H.323 (some) Working knowledge of networking interoperability, VOIP security, certificates Working knowledge of Windows Server environments Working knowledge of MS SQL (nice to have not essential) Good customer facing communication skills Good written and spoken english Must be authorized to live and work in the United States for any employer. Must be able to travel - up to 50-60% Must be able to work from home when not traveling.
Category: Networking & Security       
Preferred Location: North America

C****ive
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| C****ive
|    United States
Fixed Price: $25 - $30   |  Posted: May 22, 2015  |  Ends: 12d, 18h  |   5 Proposals
Hi i need an assistance with elastix 3 behind a router the extension dont register and i have sip trunk not working Must have an experience with elastix 3 cheers
Category: Networking & Security       
Skills: Asterisk, Cisco, VOIP Administration, Elastix       

r****raw
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| r****raw
|    Canada
Fixed Price: Less than $500   |  Posted: May 22, 2015  |  Ends: 12d, 9h  |   2 Proposals
The freelancer is required to search to modify open source java client of his choice (as long as it works on windows). the modification includes changing the user interface, removing some options, and adding some other options. A detailed document will be sent to the shortlisted candidates. The softphone should work with generic SIP PBX such as Free PBX
Category: Software Application       
Skills: Java, VOIP Software       

R****i02
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| R****i02
|    Malaysia
Fixed Price: $500 - $1,000   |  Posted: May 21, 2015  |  Ends: 6h, 19m  |   8 Proposals
Looking for someone to create an OPENSIPS switch on the Amazon Cloud Platform. It has to be able to integrate with Jerasoft billing and Dynamic routing. The switch will totally be a "dumb" switch and only needs to redirect and terminate calls. It need not be anything fancy. To see what documentation there is read   [obscured]  /display/IM/JeraSoft+VCS+Common
Category: Other IT & Programming       
Skills: OpenSIPS       

a****cha
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| a****cha
|    Hong Kong SAR, PRC
Hourly Rate: $30 - $40 / hr   |  Duration: Not Sure  |  Posted: May 21, 2015  |  Ends: 12d, 4h  |   9 Proposals
I need my Twilio SIP account attached to Vicidial for Termination and Origination (Outbound and Inbound). Please read both the Vicidial documentation and SIP documentation on this before applying.   [obscured]  /docs/sip-trunking/sample-configuration   [obscured]  /store.php(The free version) If all goes well you will be considered to maintain Vicidial for us on a per case basis.
Category: System Administration       
Skills: Asterisk, Twilio API, FreePBX, SIP, VICIDIAL       

d****dm_
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| d****dm_
|    United States
Fixed Price: $750 - $2,000   |  Posted: May 21, 2015  |  Ends: 4d, 0h  |   2 Proposals
Deliverable: Web page copy for SEO Project scope: 1) Use or edit an existing page template (link below) that describes how our product can be used to test any application that uses protocol 'X' (name of protocol to be inserted later). 2) Select a protocol from the list below and do a light research of it (Wikipedia, etc.). Estimated time for this is about 10 minutes. 3) Starting with the template, make sufficient additions or changes to 'personalize' it for the selected protocol, keeping in mind the basics of SEO. Also create a URL name, page title and meta description (all of which factor into SEO). 4) Select the next protocol, research it and again starting with the template make a page for it. 5) Repeat until there is a page for each protocol. Template page link:   [obscured]  /dynamic_fuzzing_testing_hsrp_protocol List of 118 protocols, below. Project goal: Web page, URL, title and meta description copy that is SEO friendly for each of ...
Category: Search Engine Marketing       

B****ent
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| B****ent
|    United States
Fixed Price: Less than $500   |  Posted: May 21, 2015  |  Closed  |   4 Proposals
Hi, We are looking for a free lancer who is capable to modify our existing proxy server application code in OpenSIPS. You may need to change our existing functional architecture also depends on the issues that is what we are facing (like unable to receive incoming calls in some network environments etc.). Please apply, if you have really excellent expertise and experience in OpenSIPS. Note that 1) The selected freelancer must be in online either in GTalk or Skype 2) As we have very tight dead lines, you must follow committed dead lines. 3) Payment will be done, once we test the application thoroughly. Thank you. Chandra.
Category: Software Application       

r****r12
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| Client
|    United States
Fixed Price: Not Sure   |  Posted: May 21, 2015  |  Ends: 11d, 20h  |   15 Proposals
We want to develop an instant chatting and p2p audio/video App to work on iphone mobile platform. It must work from behind the NAT (using ICE, TURN, etc.). Indeed, it should be worked in the presence of lots of internet filters. Voice/video quality management including silence detection and packet loss management. The ability to adjust Video chat resolutions. Suitable codec in order to use the app with 256KB speed. Negotiating services quality depending on connection and other factors. Ensure Jitter buffer and Echo issues are not there. For back-end, we prefer to use node.js and mongoDB. The delivered result should contain: 1. Source code with nice layout and enough comments that is easy to maintain 2. A short document to describe your code and the structure We'll get an in-house QA to test the apps after finishing them completely. Please PM me with the following information when you bid: 1. A brief introduction about you 2. Technical details of instant chat/massing technologies you a...
Category: Mobile Applications       

f****uzi
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| f****uzi
|    Germany
Fixed Price: Not Sure   |  Posted: May 21, 2015  |  Ends: 11d, 16h  |   4 Proposals
Hi I need someone to modify the data structure in Sofia-SIP library to provide better performance. Please let me know if you may have experience on this library.
Category: Software Application       
Skills: C, Algorithm Development, Algorithms       

a****omp
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| a****omp
|    Hong Kong SAR, PRC
Fixed Price: Less than $500   |  Posted: May 19, 2015  |  Ends: 10d, 3h  |   4 Proposals
Following are the basic things I need , we can discuss if you any any more questions. Please do not waste my time if you do not have enough experience in SIP and mobile app development. As for your information there is already Build branded version is available on csipsimple code repository. 1. set up an environment to build CSipSimple in Windows environment, 2. You must be able to provide a simple document of all steps along with actual setup including installations of any SDK and IDE etc ( preferably Android studio). 3. Must use latest nightly build code from svn 4. Setup Emulator on PC to test code( I know its part of android studio) 5. Few menu icons changes ( I will provide them)- hardcode sip proxy inaddition to already existing menu proxy setting - hardcode 60 day expiration. 6. Ability to enable and disable in code by setting flags during compile " READ_LOGS" and WRITE_EXTERNAL_STORAGE. see this link for clarification   [obscured]  /p/csipsimple/wiki/...
Category: Mobile Applications       

U****AMT
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| U****AMT
|    United States
Fixed Price: Not Sure   |  Posted: May 19, 2015  |  Closed  |   11 Proposals
We are having difficulty setting up an asterisk server with a public IP to work with sipML over the internet (assume sipML client behind NAT). We use DigitalOcean as our VPS but are open to other providers. We would just like a simple reference asterisk set up on our server that will work with webrtc connections (sipml, sip.js, etc).
Category: Other IT & Programming       
Skills: Asterisk, Freeswitch, WebRTC       

a****iff
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| a****iff
|    United States
Fixed Price: Less than $500   |  Posted: May 19, 2015  |  Ends: 9d, 18h  |   5 Proposals
We are looking for candidate who is having experience in developing android application. Project Description: Need to develop SIP Mobile dialer with authentication of OP code and user name and password . Job Description: We are looking for a talented Mobile Application Developer who will be responsible for the design, development and deployment of our mobile apps. Your responsibilities: - Review and perform a technical analysis of requirements - Produce a solid, detailed technical design - Write clean, modular, robust code to implement the desired requirements - Contribute ideas for making the application(s) better and easier to use Your qualifications: - A work style that is extremely detail oriented - A complete Elance profile - References or an established Elance reputation preferred
Category: Mobile Applications       

a****ben
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| a****ben
|    Oman
Fixed Price: Not Sure   |  Posted: May 18, 2015  |  Ends: 8d, 9h  |   18 Proposals
Technical Details we want to develop an App to work on android mobile platforms. It will use SIP protocol. We are looking for the following features and functionality. Developer should focus on each and every detailed information and submit the products with all mentioned details. Important Features App must be compatible with major SIP soft switches and should support all android versions. PUSH instantaneous messages (incoming calls, text messages, voice mail, new contact joined) It must work from behind the NAT Account to Account free calls (voice and video) Instant Messenger features (Account to Account free chat) File sharing(Audio , video image ect.) App should display country rate when dial a number. Add payment option feature through credit card and PayPal Enhanced Address Book Real time notification when a new user from Contacts joins the service Search users by email, service ID and name It must have dial, redial, receive, hang up, mute, speaker phone, volume cont...
Category: Mobile Applications       
Skills: Android, SIP, VOIP Software, Software development       

D****Tel
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| D****Tel
|    United States
Fixed Price: Not Sure   |  Posted: May 18, 2015  |  Ends: 8d, 8h  |   1 Proposal
We are looking for someone to create a simple meta-layer containing some recipes to build and cross compile Certified Asterisk 13.1 LTS into Yocto Dizzy (target system : genericx86-64 / genericx86 / qemux86 / arm) Your responsibilities: - Create the needed recipes - Asterisk has to be built from original sources (  [obscured]  /pub/telephony/certified-asterisk/...) - Create recipe for unavailable dependencies - Configure Asterisk to support SIP - Test the result Your qualifications: - Good knowledge about Yocto and Open Embedded - Basic knowledge on Asterisk PBX Available layers dependencies : - poky/meta - poky/meta-yocto - poky/meta-yocto-bsp - meta-oe/meta-oe
Category: Other IT & Programming       
Skills: Asterisk, Yocto / Open Embedded       

s****net
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| s****net
|    Switzerland
Hourly Rate: Less than $10 / hr   |  Duration: 1-2 weeks  |  Posted: May 15, 2015  |  Ends: 5d, 19h  |   14 Proposals
We are seeking a freelancer to help build our list by identifying qualified businesses for our services by doing the following: * make calls using our VoIP (SIP) server, using their own hardware or software phone, and account credentials we supply. * follow scripts we provide. * make calls Monday through Friday, between 9AM to 5PM, Eastern time (UTC?05:00), excluding major holidays (which we will identify in advance). * update information as calls are made in a Google Sheet we provide to capture information obtained. * speak and make all calls in US English. * make at least 40 dials per hour in attempts to reach prospects. All calls will be recorded. Prior to being awarded this job, Freelancer is expected to call and leave us a voicemail, reading the following script so that we can assure Freelancer is well-spoken and coherent. The number will be given to those who contact us, requesting it, prior to bidding on the job. =================================================...
Category: Telemarketing       
Skills: Lead Generation, Telemarketing       

p****ern
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| p****ern
|    United States
Hourly Rate: $40 - $50 / hr   |  Duration: Not Sure  |  Posted: May 14, 2015  |  Closed  |   12 Proposals
Voip Setup of a loaction, onsite or remote with an onsite technician. Setup a Cisco2911 Router for connectivity of 5 lines 15 phones, need to be able to configure SIP to get site operational, with voice mail and call forwarding.
Category: Networking & Security       
Skills: Cisco, VOIP Administration, .NET Remoting       

I****ide
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| I****ide
|    United States
Fixed Price: $1,000 - $5,000   |  Posted: May 14, 2015  |  Ends: 4d, 12h  |   16 Proposals
He should be familiar and comfortable with Asterisk installation, configuration, Dialplan programming and troubleshooting. Preference would be given to having experience on IVR and Call center application development. SIP and protocols Asterisk integration with PSTN is preferred. Asterisk integration with Database (MySQL and MSSQL) with knowledge of queries and stored procedures. Scripting with Shell, Perl Candidate Profile The candidate must have: - minimum 1 years of experience working on Asterisk platform. - sound knowledge of Asteirsk Dialplan, Asterisk Gateway Interface and Asterisk Manager Interface. - worked on Queue, IVR and Voicemail related applications. - hands on experience of installing and configuring Asterisk with PSTN card. - R & D skills in Asterisk. Nice to have: - good knowledge of SIP and IAX. - basic knowledge of Linux commands, MySQL and Telephony Domain. - ability to test and troubleshoot Asterisk applications/issues - Knowledge of C programming in As...
Category: Software Application       
Skills: Asterisk, PHP       

h****000
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| h****000
|    India
Fixed Price: Less than $500   |  Posted: May 13, 2015  |  Closed  |   13 Proposals
Hello This is to customize the LinPhone app and do the following: 1. Change the apps name 2. Change the apps icon 3. Change the main landing page 4. change sip server default name 5. Change about page (where required) 6. Hide some items on the settings page Thanks Dominic
Category: Mobile Applications       

****
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| Client
|    United States
Fixed Price: $50 - $100   |  Posted: May 13, 2015  |  Closed  |   3 Proposals
Hey there! I have a OpenCom 100 with an extension set, 1 x 4 UPN DECT Modul M100-U4d and 1 x Modul M100 8UPN DECT OpenCom and one RFP 21 (DECT Station). I also bought on upgrade license OC130 and one Licence OpenLine SIP 2 Telephony. Because I am not a expert in this stuff I search for one :-) You need to do: - generate and install both licence keys - setup the VOIP (4 voip accounts) Thats it. You will get access by teamviewer. Thanks for your help!
Category: Other IT & Programming       
Skills: VOIP Administration, VOIP Software, aastra, mitel       

s****reg
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| s****reg
|    Switzerland
Fixed Price: Less than $500   |  Posted: May 13, 2015  |  Ends: 3d, 19h  |   3 Proposals
Have Asterisk 1.8. Currently using as an Auto Dialer. We will have 3 agents on standby connected to SIP phones on same PBX. Need Asterisk modified so when a an auto dialed customer answers, our agents in our call queue get that person on their SIP phones.
Category: Other - Administrative...       
Skills: Asterisk, VOIP Administration, VOIP Software       

g****ent
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| g****ent
|    United States
Fixed Price: Less than $500   |  Posted: May 13, 2015  |  Ends: 3d, 13h  |   3 Proposals
need help configuring/customizing oversip to run on PC, MAC, IPHONE, PADs the web service must work with most browsers talking to freeswtich
Category: Software Application       
Skills: Web Services Development, SIP       

t****n61
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| t****n61
|    United States
Hourly Rate: Less than $10 / hr   |  Duration: Not Sure  |  Posted: May 13, 2015  |  Ends: 3d, 6h  |   14 Proposals
I have already setup goautodial server but need someone to setup and configure VICIDIAL and sip trunk.
Category: Software Application       
Skills: Asterisk, PHP, VICIDIAL       

i****011
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| i****011
|    Australia
Fixed Price: Less than $500   |  Posted: May 12, 2015  |  Ends: 2d, 14h  |   6 Proposals
Linphone sdk integration with android application. There is an existing android application. The programmer needs to integrate linphone sdk with this app. The functionality is simple: The application has a unique uuid, the apps should be able to use this uuid to get sip number and secret from sip server and configure it in the app. When a call is received on the app, then pull some data from the remote server and display it while the phone rings or the call is underway.
Category: Mobile Applications       

a****001
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| a****001
|    India
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