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Asterisk Jobs

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Hourly - Expert ($$$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We have developed a system in FreePBX/Asterisk to stream audio to a caller, using the Music On Hold system. ... The bitrate of the audio is low however, leading to poor quality when the call is made into the system. I believe this is because the MOH module in asterisk is limiting the stream to 8khz We would like to change this, so we are able to stream higher quality audio to callers.
Skills: Asterisk FreePBX VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $40 - Posted
Running ViciDial system (on OpenSuse, with Asterisk 1.8, etc) Looking for someone experienced with ViciDial and creating webphones. ... Most likely did step 7 wrong as I'm not familiar with setting up web based applications. I need someone good in asterisk to see the issue, fix the issue and report step by step how the fix was applied.
Skills: Asterisk HTML5 openSUSE vicidial
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Hello, I am looking for a vicidial Avatar/Soundboard that someone has developed to link into my current phone cluster. If you have developed an avatar type system for vicidial please send me a message.
Skills: Asterisk PHP vicidial VOIP Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
Need current Asterisk demo environment extended to use Kamailio for sip registration of two phones and one sip trunk. ... Freelancer will be responsible for taking a working Asterisk configuration (sip.conf, res_digium_phone.conf, extensions.conf, and voicemail.conf + any other freelancer need deems necessary) and helping provide us the ability for sip registration to be moved to Kamailio while utilizing Asterisk for backend media. ... Freelancer will be responsible for taking a working Asterisk configuration (sip.conf, res_digium_phone.conf, extensions.conf, and voicemail.conf + any other freelancer need deems necessary) and helping provide us the ability for sip registration to be moved to Kamailio while utilizing Asterisk for backend media. Would also like to continue to register the phones (Digium D45 and D70s) utilizing res_digium_phone.conf file in asterisk if possible.
Skills: Asterisk OpenSIPS Ubuntu
Hourly - Intermediate ($$) - Est. Time: More than 6 months, 10-30 hrs/week - Posted
We have an Elastix installation (with Openfire and HylaFax), which needs maintenance regularly. We are looking for someone who is able and willing to do the following: - maintenance of the system (OS and software components) - additional feature setup/programming/changes We are looking at a long-term relationship.
Skills: Asterisk
Fixed-Price - Intermediate ($$) - Est. Budget: $250 - Posted
We have existing Asterisk setup which supports our internal SIP calls. We'd like to add: - Incoming calls with voice menu (actually we have incoming number, but it is not in use) - Voice conferences - Video calls - Chat We need a contractor who can configure our Asterisk this way keeping UI alive because we use it for everyday administration of users and trunks. ... We'd like to add: - Incoming calls with voice menu (actually we have incoming number, but it is not in use) - Voice conferences - Video calls - Chat We need a contractor who can configure our Asterisk this way keeping UI alive because we use it for everyday administration of users and trunks.
Skills: Asterisk Linux System Administration SIP Ubuntu
Fixed-Price - Intermediate ($$) - Est. Budget: $300 - Posted
When the caller dials our number, he should be prompted by IVR to key in 8 digits and he will be placed in wait queue. When the agent picks the caller from the queue, she should have her browser automatically display the detail view corresponding to a sugarcrm lead , where the name field ends with 8 digits. So if the caller punches in 12345678 in the prompt, then lead having name “FP12345678” should be displayed in the agents browser. Example: lead object with name FP12345678 has id 757db759-0823-586a-3f69-559ba018f52a browser should show: /index.php?module=leads&action=DetailView&record=757db759-0823-586a-3f69-559ba018f52a If the caller doesnt punch in a valid number, or if there are multiple matches, the call should still get connected and browser should show index page of object: /index.php?module=lead&action=index Note: I can provide both elastix and sugarcrm test interfaces
Skills: Asterisk Elastix FreePBX PHP
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