Asterisk PBX Jobs and ContestsAsterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
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|PBX IP Trunk||Create a voip trunk with two server 1. FreePBX , 2. Thirdlane||4||Asterisk PBX, VoIP||Mar 27, 2017||Today14h 8m||$26|
|Viber Call to Asterisk||Hello There, We are looking from someone who can integrate Viber call s to Asterisk, Were we need any one who calls us on our Viber number to be connected to our Asterisk. Very Simple you are free to choose the way you want to implement this task, like sip trunk or any other tool you may need||0||Asterisk PBX||Mar 27, 2017||Today6d 14h||-|
|Solve SIP Kamailio Problem||Expert kamailio only||4||Asterisk PBX||Mar 27, 2017||Today6d 10h||$204|
|Ubuntu package Update Prob||i want to update my ubuntu package useing command apt-get update its show error main package 404 not found i just want to update package and install asterisk & openvpn if you can help to to fixed package issu please bid||30||PHP, Linux, Asterisk PBX, VoIP, Ubuntu||Mar 26, 2017||Mar 26, 20175d 15h||$27|
|Business VoIP Service U.S. Leads||We are looking for Business VoIP service lead generators. Leads must Come From U.S. Companies. $50 for Confirmed Lead. Contact must be willing to speak to us about their VoIP Phone service. We will pay based on weekly successful leads provided. Thank you||8||Sales, Bulk Marketing, Asterisk PBX, Leads, VoIP||Mar 25, 2017||Mar 25, 20174d 15h||$768|
|VoIP Customer Leads||We are looking for Business VoIP service lead generators. $50 for Confirmed Lead. Contact must be willing to speak to us about their VoIP Phone service. We will pay based on weekly successful leads provided. Thank you||4||Sales, Asterisk PBX, Leads, VoIP||Mar 25, 2017||Mar 25, 20174d 15h||$748|
|Configure Elastix - Asterisk with IPTABLES - FIREWALL: only Inbound 20 IP and outbound 4||I have one server of Elastix and i want to enter 20 IP that i have one list, and 4 outbound IP. All ports have close and the server have to secure.||13||PHP, System Admin, Linux, Asterisk PBX, Network Administration||Mar 25, 2017||Mar 25, 20174d 9h||$37|
|free pbx||Currently have a working FreePBX install on Sangoma hardware. The hardware is old and not rack mounted. I have new rackmount hardware (online, ready, with IP address, free PBX installed etc) the following needs to happen to it; Port the current working config to the new hardware. Re-configure the phones to point to the new hardware. The phones are Yealink T28P phones On the handsets, make the BLF fields light up as if their line keys (Line1 Line2 etc - 1 per SIP trunk) - i have 5 phones, if you can make one work i'm sure I can copy the config. Configure the GSM card (Sangoma W400) - The new hardware has a 2 port GSM module in it, which will be used for voice and sms, so configure the SMTP reciever and the voice sims as a trunk etc. Configure the 2 analogue trunks & 2 analogue extensions (Sangoma A400 card) so they appear in the gui. Thats it!||8||Asterisk PBX, VoIP, FreeSwitch||Mar 24, 2017||Mar 24, 20173d 9h||$163|
|I would like to hire a Cisco Engineer||I have a Cisco UC520 that needs to troubleshooting support. I can not figure out why an inbound call keeps dropping.||15||Asterisk PBX, Cisco, VMware, FreeSwitch||Mar 23, 2017||Mar 23, 20172d 22h||$204|
|Setup AutomaticDialer playing a Sound (Wav File) Elastix||I need setup module with: 1.- Your upload a CSV File or XLS File with Phone Numbers to call 2.- You uploada Wav File for playing 3.- You clic RUN and Elastix will be call to the numbers on XLS or CSV File playing the wav I need to do in exactly 2 hours remotly||3||Linux, Asterisk PBX, VoIP, MySQL, UNIX||Mar 23, 2017||Mar 23, 20172d 9h||$63|
|Asterisk Auto Dialer Application||Create Application or Web based app get list of users from database call users on order log call status (success,failed,answered, action taken) press key send http request set number of simultaneous calls||10||Asterisk PBX||Mar 23, 2017||Mar 23, 20172d 2h||$158|
|Make some changes on Asterisk server||Minor changes including: - Adding a recording if the user doesnt dial the right number on their phone. - Letting the user try again. -If no # dialed on the phone and hung up return a 0 to the php based dasboard.||6||PHP, Software Architecture, Asterisk PBX||Mar 22, 2017||Mar 22, 20171d 20h||$33|
|I would like to hire a Cisco Engineer||We occasionally have a need to configure Cisco hardware and it isn't a core competency of ours.||35||Asterisk PBX, Cisco, VMware, FreeSwitch||Mar 22, 2017||Mar 22, 20171d 18h||$425|
|Asterisk 13 SMS||I have "Asterisk 13.1.0~[url removed, login to view]" its been working fine as my private voip server. However I would like to use the sms / messaging feature. I do not have any GUI or 3rd party addons this is a stock Ubuntu Asterisk install. I am using Zoiper on Android and would like to use the SMS / messaging feature. Also my DID supports incoming/outgoing SMS so would like that to work. I would like if possible some one to guild me through what I need to do to make this happen. I cant give access to the server but can give the files u request (less any private info in them) Please dont just point me to some readme u googled as I have tried a few with out luck. So this will most likely need some one with experience.||9||Linux, Asterisk PBX, VoIP||Mar 22, 2017||Mar 22, 20171d||$12|
|GoSub Error in Asterisk||I have a voice blast software with code written in asterisk and Php. It is integrated with an application and on hangup of the call, i am sending a hangup message to the application. The code functionality is working fine except in one scenario. While the call is been executing, if the customer disconnects and it moves to GoSub return context , in this particular scenario rest of the dial plans are not executing. It will hang up the local channel [url removed, login to view] log for SIP/Dahdi channel hangup so i am not able to trigger the hangup message to the application. It is an urgent requirement to be started today. Only bid if you are an expert on asterisk dial plan.||5||PHP, Asterisk PBX, MySQL||Mar 21, 2017||Mar 21, 201723h 43m||$38|
|Fixing bugs to Asterisk Panel||Hi, We have a Live Asterisk panel that need bug fixing.||22||PHP, Software Architecture, Asterisk PBX, MySQL, Bootstrap||Mar 21, 2017||Mar 21, 20173h 43m||$165|
|Project for voiplinux||Hi voiplinux, I noticed your profile and would like to offer you my project. We can discuss any details over chat.||5||Linux, Asterisk PBX, Technical Support, VoIP, , Call Center||Mar 20, 2017||Mar 20, 20172d||$26|
|Project for voiplinux||Hi voiplinux, I noticed your profile and would like to discuss a potential project..||2||Linux, Asterisk PBX, Technical Support, VoIP, , Call Center||Mar 19, 2017||Mar 19, 20171d 17h||$12|
|Convert PC-based VOIP Soft-switch client to APP||1) VOS is a PC based client-server soft-switch. It provides a client running on Windows to connect to the Linux based soft-switch. We want to develop the client into App which can be run on iOS and Android. 2) VOS provides an API which provides all the functions necessary to develop App. 3) The developer is required to study VOS functions and use VOS API to develop App. 4) The developer must have 5+ years experience in VOIP/telecom field, with in-depth knowledge on soft-switches and inter-connection, mapping, routing, billing etc. 5) The developer must possess good self-learning ability who can read and understand technical documents, and develop new applications based on user-manual provided. Only experienced developer(s) in soft-switch will be considered. Please don't apply if you don't know soft-switch.||28||Mobile Phone, iPhone, Asterisk PBX, VoIP, FreeSwitch||Mar 13, 2017||Mar 13, 20171d 23h||$671|
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