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Asterisk Jobs

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Fixed-Price - Est. Budget: $ 10,000 Posted
Hello: I'm starting a VoIP company, I need an experienced professional who can efficiently and swiftly set up a complete system. The system must be able to: -Users can make accounts on their own -Set up a text message chats -allow for video conference -allow for file transfer -allow to call land lines where users can add credits using credit card or paypal By all means a system comparable to Skype or Viber In terms of which system protocols to use i am open to most, lets discuss them if you choose to accept. If you have hardware experience that is a plus, as I would like to scale up this and need to know the hardware thats best. Also, Price is not Fixed and we can discuss pricing Whether you want to do more or less in the scope.
Fixed-Price - Est. Budget: $ 200 Posted
Hi there, We are looking for an experienced highly skilled developer who re-writes the responsive GUI with Admin and agent interface for VICIdial utilizing Bootstrap, CSS, PHP and AJAX. The design should also be fully tested against the major browsers, particular Firefox & IE and optimized for all monitor sizes. Prospective developer would have in-depth knowledge of PHP, Mysql, Perl, CSS and AJAX, Asterisk, VICIdial, Linux. To qualify, please send us screenshots of past work with the ViciDial Admin and Agent UI.
Fixed-Price - Est. Budget: $ 600 Posted
Hi, We are looking for a Sugarcrm-Asterisk integration with the below features. We'd prefer a ready made product with minor customizations needed. The solution should have the below features: 1. Integration between Sugarcrm CE and Asterisk 2. Solution should be upgrade proof 3. Click to call a. Contact any SugarCRM client directly from the CRM record with the simple click of a button 4. Lead/Contact preview popup a. client records will automatically be displayed prior to any incoming call or outbound campaign, allowing representatives to immediately view all relevant customer data to increase first call resolutions or improve every sales and marketing opportunity. b. The popup should also display the notes from the previous call (if any). This should work for inbound as well as outbound calls both c. We’d like this popup to display a custom field too 5. Call Recordings and notes a. Notes for the call can be saved from popup during ongoing call. b. Automatically record...
Fixed-Price - Est. Budget: $ 60 Posted
I wish to hire someone with good communication skills who can show me how to do a few things on my Grandstream UCM6102. I am quite knowledgable in IT in general - I just don’t have time to play around with the UCM6102 and discover how to set the device up myself. I expect a well prepared tutoring session over Skype and Teamviewer. I foresee the initial part of this contract to involve up to 2 hours tutorial. However, I imagine requesting the support of the VoIP expert sporadically in the future when troubleshooting issues or wishing to add new features. As the UCM6102 can only be accessed on our local network, I am happy to let the tutor teamviewer into my computer and look around the UCM6102 configuration before delivering to me the tutorial. What I would like to set up: Path 1 - office hours (Mon-Fri, 10am-10:45pm; Sat, closed; Sunday 8:30am-6pm) When someone calls, a welcome message played to everyone (OHGreeting). Then all phones ring (Hold). If not picked up, goes to...
Fixed-Price - Est. Budget: $ 150 Posted
1. Setup the security and call quality settings correctly. 2. isymphony complete setup 3. incoming call diversion to operator logging, (may have 2 operators) 4. correctly setup/verify approx. 10 extensions and 2 pstn connection 5. call recording schedule into dropbox or Justhost folder 6. make sure pabx run smoothly without having any security issues 7. change ivr menu options and hold on music options 8. setup on demand recording individual files to as user option to email or downloaded 9. help to setup headset’s as optimum use conditions & rectify current issues with existing trunks 10. Given full tanning including; Free PABX commands essentials Setup and use of isymphony Setup new extentions call recording Managing backup’s
Fixed-Price - Est. Budget: $ 100 Posted
Tehnology we use: -Asterisk -FreePBX -WebRTC (Tried with Sip5ML, JsSip, Sip.js) - 4 different trunks - One stationary number (our ISP) - 4 GSM numbers Problem we have: When calling trough stationary trunk on stationary trunk we get sound delay when answering late (Delay is not always the same it depends on the time when user pick up the call) Explanation: Ok so we start off with what is working. If we make a phone call through GSM trunks on any number (GSM or Stationary) everything works fine. The important thing here is that when connection is established and call starts to progress we get response 183 (With early media) and so on... The 183 response is sent by our GSM VoIP terminal. The problems appear when calling through our ISP trunk, since provider can't fake 183 response if it is not sent to them from dialed number. When we call through ISP trunk on GSM numbers things work fine, when connection is established GSM providers send 183 response with early media...
Fixed-Price - Est. Budget: $ 200 Posted
Hi I need help with eSpeak TTS, i am having issue to play it via Asterisk, if you have experience with this , then please reply to this job posting Thanks Aron
Skills: Asterisk
Fixed-Price - Est. Budget: $ 200 Posted
We are using Asterisk 1.6.2.9 on a Ubuntu2.1 server with Yealink T28 handsets. We are experiencing intermittent drop outs on incoming calls (outgoing calls never drop out). We probably get on average 20 incoming calls a day and approximately 2 calls would drop up within 2 to 3 minutes of the call being answered. We are using a Digium Asterisk TDM411B card for incoming calls from an existing POTs number and also have a virtual number from Voxbone. Both numbers seem to be experiencing this incoming drop out problem. What we need you to do? 1) We need someone to turn on all debugging on asterisk to find out what is causing the drop outs on these incoming calls and fix it. 2) Once debugging is turned on we will tell you the phone numbers of the incoming calls that drop out each day so that you can trouble shoot and fix the problem. We know that it is not the handset or router that is causing the drop out as we have changed the handset and router and still have the same problems. Upgrading...
Skills: Asterisk