Asterisk PBX Jobs
Asterisk 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.
Hire Asterisk PBX DevelopersWe need to build our own SIP/ VoIP/ tfn server to setup the toll free numbers to sale the client.
Call center required for calling 1000 numbers with IVR recording and quality
Hi, I am looking for someone who can Install / Configure Asterisk based Free PBX 1. I want to use GSM gateway 2. Click to call API 3. Condition based IVR
I am looking to hire someone whom can prepare instructions for configuration the EdgeMarc 4806. We have many units to deploy and are looking for someone who can prepare a configuration guide. The guide must cover steps to configure the SIP-UA for the FXS ports and connect them to Hosted PBX service, as well as how to configure the T1 interface as a Network-side PRI interface for a fractional (8-...
Hello Guys, I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP [login to view URL];branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@[login to view URL] From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:0...
I enabled SonicWALL VPN and i used for NEC IP phone but SonicWALL is blocking the packets so it is ringing but no sound coming
We have a working 3CX server. All phones and lines are working OK. We have the 3cx PRO version. We need to makean IVR menu (see attachment) We will provide all mp3 files. Ignore green letters on atatchment (this is the text for the mp3 files we will provide)
The Project requires a developer with knowledge in the VoIP industry and utilizing Vue and node.js the web App will deal with carrier APIs ( Like Bandwidth ) grab the services, like SIP TRUNK, Number ( DID ) selling, SMS Selling, Voice and e911 reselling) the user will subscribe to our portal, select the product ( e.g Buy Phone Number ( DID ) will go through filtering by selecting Country, City, S...
I need someone to assist us in making outbound calls to potential sellers. You must be willing to work for $0.50 per hour between the hours of 8 am - 9 pm eastern time. I’m looking for an average of 800 to 1000 calls per day. I will provide autodialer. Will be trained and given scripts. Must speak fluent english
I am looking to develop API for Grand stream / Open Vox IP PBX system using the Asterisk Manager Interface (AMI) or using the API document which is given by the Grand Stream (find enclosed) I need the following API Click to Call API Auto call forwarding to extension based on the mobile number Getting call logs data - CDR / CLI
Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Budget $50 Thank you.
We would like to create a Voice Modulation that takes normal voice as input and modulates it into funny voice and then provides the voice as output. Similar to this [login to view URL]
WE ARE WORKING TO BUILD CLOUD-BASED HOSTED IVR SERVICES AND FOR THAT WE ARE LOOKING FOR A DEVELOPER AND SOMEONE WHO CAN BUILD THE USER INTERFACE TO BE USED BY OUR CUSTOMERS AND DO ALL THE TECHNICAL WORK THAT IS REQUIRED TO BE DONE WE WILL BE STORING ALL THE DATA AND CALL RECORDING ON DATA CENTERS WHICH OUR CUSTOMERS CAN ACCESS VIA API
Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wake...
NEED HELP TO SETUP VOIP SERVER - RUNS ASTERISK AND ALSO ONGOING SUPPORT
Need help to setup billing based on the package on my website. My website is VOIP credit selling. Let me know if you have experience.
I do have VOS3000 2.1.2.4 Security Authentication Panel installed on my server, I need help to configure it so that it will automatically block the SIP IP Address that has 3 failed login attempt. Also the Admin should be able to unblock the IP Address once the it has been verified that IP Address in question is not a hacker!
Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Thank you.
I am looking for softphone for Windows just like linphone. Let me know if you have any questions.
A dynamic regional Chicago-based Unified Communications services provider is seeking to engage an individual who: 1. Understands Software Product Development Lifecycle 2. Understands developing continuous integration and continuous 3. Capable of working independently in a small team This is a long-term ongoing full-time engagement. Responsibilities include: • Application and Infrastructu...
Setup Monitoring with Zabbix 5.4 for an Old AsterixPBX (v11) (With Videotutorial for own use)
need Jitsi and jibri expert. installed jitsi and jibri servers and meetings and screen sharing and other features are working well. but only recording is not working. checked all instance configures and server configures and it looks like correct. Jibri server connects xmpp server well.
Creat FreePBX on pc and make it work offline and online on the network + configuration for Grandstream GXW42XX Gateways and GSM gateway and other configuration
I recently installed a new FreePBX on the cloud and purchased the Fax PRO module, it works fine for receiving fax but when sending out fax the following error appears: The call dropped prematurely
Seeing failures in the field at NAND flash from bit flipping. Require a low level linux engineer with extensive knowledge on NAND flash and ideally corewind module as well. Issue is with Corewind module and Micro 8 bit controller Issue is with bit flipping and bad sectors causing our product to fail in the field at rate around 1%
I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP [login to view URL];branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@[login to view URL] From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:046022600@xxxxx...
