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 DevelopersI need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not ...
Need a support for existing zabbix server fine tune & configure two proxy server.
[login to view URL] In this CRM add the VoIP API integration for calling feature. inbound/outbound feature. Remove back links I will give u VoIP API.
[login to view URL] In this CRM add the VoIP API integration for calling feature. inbound/outbound feature. Remove back links.
[login to view URL] In this CRM add the VoIP API integration for calling feature. inbound/outbound feature. Remove back links
We are looking for Customer Service Representatives to join our team. This is a remote position. Customer Service Representative Job Duties includes, but not limited to: ● Handle in/outbound calls and e-mails. ● Execute excellent customer service via chat support with online customers ● Resolve product or service problems, queries and negative feedbacks in a positive and professional way at al...
[login to view URL] In this CRM add the VoIP API integration for calling feature. inbound/outbound feature.
We are looking for Customer Service Representatives to join our team. This is a remote position. Customer Service Representative Job Duties includes, but not limited to: ● Handle in/outbound calls and e-mails. ● Execute excellent customer service via chat support with online customers ● Resolve product or service problems, queries and negative feedbacks in a positive and professional way at al...
We are looking for Customer Service Representatives to join our team. This is a remote position. Customer Service Representative Job Duties includes, but not limited to: ● Handle in/outbound calls and e-mails. ● Execute excellent customer service via chat support with online customers ● Resolve product or service problems, queries and negative feedbacks in a positive and professional way at al...
We are looking for Customer Service Representatives to join our team. This is a remote position. Customer Service Representative Job Duties includes, but not limited to: ● Handle in/outbound calls and e-mails. ● Execute excellent customer service via chat support with online customers ● Resolve product or service problems, queries and negative feedbacks in a positive and professional way at al...
We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a pub...
Hi I need CRM with calling feature.. I will give VoIP API..
We are looking for Customer Service Representatives to join our team. This is a remote position. Customer Service Representative Job Duties includes, but not limited to: ● Handle in/outbound calls and e-mails. ● Execute excellent customer service via chat support with online customers ● Resolve product or service problems, queries and negative feedbacks in a positive and professional way at all ...
Hello there, I have got an VM WARE ESXI in a dedicated server, and i want you to install goautodial or Vicidial on it. So that i can log in directly to goautodial by just putting the ip on the url from anywhere.
Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database
I have a Vmware ESXI installed on a dedicated server. I have installed vicidial (goautodial) as well. But not able to set it up with a static ip.
Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall...
I have a running asterisk PBX - i will to rebuild new one with asterisk. i use Asterisk API, Databases.
- Reconfiguration of router to remove static route VPN - Reconfiguration of Server Side service to remove static route - Bridging PABX – 3cx Local/ 3cx Cloud - Creation of outbound rules – 3cx Dubai / 3cx Singapore - Moving gateways trunks to 3cx Singapore
Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action
i need someone to help me with problem that i'm facing the problem is i have linux centos and i have php file inside /var/www/html when i access from client throw browser the php files are working fine but i need when i run the php file from centos server send the result to the client browse like for example if i need session between the server and client browser
I need a VoIP Web-Dialer (not for a regular voice-call) which will dial only 1 specific number and afterward is being answered, it should send a series of DTMFs. The process is static and does not vary, "Invite -> 200 OK -> Wait almost 10 seconds - send DTMFs, wait another 10 seconds and send another DTMFs. And thats the whole process.
I have a call center that makes outbound calls using Ytel dialer system but the calls pop up as "Scam Likely" I would like to come up with a solution to be able to call people and have the number show up instead of the "Scam likely" label coming up. The Scam likely label is an algorithm when a certain number makes many calls throughout the day.
I am wanting a Server Based Block Grabber with warehouse filter for an android program called Amazon Flex. Blocks are released by their server and I am trying to be able to grab them before they get out to other users. You will need to install a proxy server onto a VPS. You will need to have this connect to the server to access the blocks for the Flex program/app. I need it to check for blocks b...
I need someone to help configure the Patton SmartNode 4114 (FXO Port) with Freepbx/Asterisk so we can use SmartNode 4114 as a VoIP gateway for PSTN lines.
i need change imei for usb huawei on raspberry with centos
I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls usi...
I had configured an ivr but the recording is not playing in Freepbx 13. I just need a help to fix it.
SiP server with low latency and push to talk features open source implementation is preferred...no closed source and I would like documentation and source code if there is android app...documentation how setting server if using open source already existing app
Have FusionPBX installed, I need to connect with SIP
I want to build a conversational IVR(interactive Voice Response) with UI with callback feature with call masking,features with voicemail and transcription of voicemail sent to theon call person and have a on call schedule
We’ve got an asterisk system with two trunks. We need some extension configuration changes and some ongoing support
I am seeking an Outbound Proxy server to be installed on one of my servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeabl...
I want to change A2billing AGI to FastAGI due to performance and scalability. I need a very experienced person with a2billing and ofcourse asterisk.
SPA400 is not deriving call to Freepbx, but it's registered, but the call drops Need help on troubleshooting and correcting the problem
CRM software with calling feature. VoIP based.
calling feature CRM. call center purpose. VoIP based.
I need calling feature CRM... VoIP based. call center purpose.. Budget is 12000 maximum.
Need some basic configuration settings done for vicidial: 1. Accessing the Agent Login through softphone on static ip 2. SMS Templates configuration on call dispositions 3. Adding dropdown menu in vicidial configuration page. 4. Leads to be recieved in the vicidial campaign via an api.
hello, we use a "less secure app" with our Asterisk PBX voicemail to email message notifications using gmail. gmail wants users who use this option to make them more secure. if you google "less secure apps" and gmail / gsuite you will see what needs to be done. this is what needs to be done: The G Suite Team <gsuite-noreply@[login to view URL]> Tue, Jul 30, 12:16 AM to m...
Integrate the vicidial sofware with the functionality to be able to push the Contact screen data, using the webform function, to a new contact screen in vtiger open source CRM. Would also need the click to call ability in vtiger.
Convert files from wav to mp3 files after a call is made, historic data and new data automatically after a call is made. when change is made I want to see and hear from crd reports. review and clean log data from /var/log/asterisk make rule to minimize log file size /var/log/asterisk/fail2ban /var/lib/fail2ban /var/lib/fail2ban/[login to view URL]
we need an expert in DIN star gsm gateway we need an expert t o set up the unit
Hello i need VoIP Call Bandwidth Optimizer / compression openWRT - SBO TP LINK & other model/ version with billing +call filter option. If u before working this or have full idea please contact with me. Thanks
we need a Din star GSM gateway to set up termination rout
i want when i send call in goip gw send voice before Early Media
Build a system that give you an virtual number which gets forwarded to another number.. MUST BE ABLE TO RECORD CONVERSATIONS
We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...
