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    11,985 asterisk opensip jobs found, pricing in USD

    Hello, I need to forward or transfer sms from smpp provider to asterisk server as sip messages and also send-back the "deliver report" to smpp provider when asterisk server terminate that message. So people with good knowledge of smpp, smsc and asterisk server can apple for this job. You can use Worker/thread on smpp database to capture values from database and then to asterisk server f...

    $136 (Avg Bid)
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    Hello I am looking someone experienced with Asterisk pbx systems who can help me to create call center on dedicated server. no newbies I will hire someone with good experiance.

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    I am the co-founder of a medical tech startup. As part of our HIPAA compliance readiness we need to create a disaster recovery and business continuity plan. This project is for the creation of this document, preferably with minimal involvement for us aside from answering questions about the business and the contingency plans we have. I prefer to use Google Docs for collaboration. The way this w...

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    We have setup ASTERISK working fine incase of callback, we want to connect agent first on call(agent should get ringing ) then will connect and then call with customer.

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    Hi, PLEASE READ CAREFULLY THE WHOLE OF THIS BEFORE BIDDING We have a project for an Asterisk IP Pabx expert to configure a hosted phone system in the cloud. We require up to 10,000 extensions eventually. These extensions are to provide emergency dial out facilities for an emergency call system for elderly people. Each elderly persons home will have a unit that looks like an IP extension off the...

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    Dear Freelance, We are looking for Asterisk and Laravel engineer to help us build a system of billing. This system will provide billing to end user. It is a customized setup and it will retrieve data from AsteriskCDR, A2billing database and FREEPBX database to create a end user billing platform.

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    IPRN SWITCH 4 days left

    Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all details:...

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    we're looking for an Asterisk developer "Should have experience in working with full Asterisk package"...if you are interested do ping me (Removed by Freelancer.com Admin)to discuss our requirement. Thank you.

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    Hello, We have a working FreePBX [login to view URL] distro installation working since 2 years. Now we want to do following enhancement in our FreePBX server. 1. We want to use our FreePBX server for internal office text message communication. I know this can be possible using XMPP module of FreePBX. But don't know, how to configure it. You have to do that. On the other hand we want to tran...

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    Hi Harmohit S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I want to setup click2call on our asterisk based Vitalpbx.

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    Hi Team, I need someone to setup click2call on our asterisk baes vitalpbx.

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    We have two fresh installations of vicidial and goautodial. We would like to configure one to test some outbound voice blast campaigns. We would like recommendations on which product to use for our needs. So Goal is: 1- initial trunk setup to asterisk or sip trunk setup. 2- Setup 2-3 campaigns based in one file txt each with all non agent "virtual/voice blast" campaigns. We have 5 differ...

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    HI would like to add an external IP when calls sent from voip switch to asterisk and add and an external ip which would be media ip and sent back the calls to Voip Minutes.

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    Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

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    Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

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    I have a grandstream model UCM 65 10 that reboots and connects the voip trunks very well. however after some minutes, the registration refresh is not working properly. need help: this is the error message in the asterisk logs WARNING[1923] chan_sip.c: Timeout on 896482223-292013357-1450912169 on non-critical invite transaction.

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    Hi I need an experienced developer with wallboard design , that understands a call center operation and wallboard requirements also reports , logon and logoff features etc. Thasnk

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    We have Installed the latest version of issabel with asterisk 16. We Also installed Vtiger CRM 7.1.0 and the connector works succesfully. At the moment click'n'call works perfectly and it's possible to pair an extension with a single vtiger profile. We are also able to see a pop up each time we call a number from vtiger interface, This pop-up only says that the extension received th...

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    I have odoo v11 community edition. I am looking for a freelancer who can help me to implement VOIP calls form Odoo. I would like to implement function of click to call in contacts.... I dont have any asterix, freePBX or cloud calls system yet, I do have SIP Trunk lines. I have seen many projects that had integrated odoo with FreePBX system, and I would like to do the same. So in short you will be ...

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    I need expert guys who have good knowlege of opensip and Asterisk

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    I am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr without rates for now. Platform must be able to handle thousands of concurrent calls and also doing rtp proxy. Platform must have a did management feature and the ability to provisioning and turn up customers immediately, retail with1 number or wholessle with thousands of ...

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    Hello, I'm running VitalPBX. Using PJSIP extensions with SIP trunks. All with TLS signalling and not encrypted RTP. When placing multiple calls at once, voice quality is horrible and choppy. I need someone who can look into Asterisk settings for me and possibly do a network capture and understand what is going on.

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    i need a sysadmin to Install A2Billing + Asterisk + FreePBX

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    Hello i want an expert to install FreePBX + Asterisk and A2Billing.

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    I want to create a Voip trading platform (like for example: [login to view URL] - See Video ) for VoIP sellers and buyers. The system has to work with Kamailio and Asterisk.

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    Need someone to configure click2dial to make calls. Calls using softphone and freepbx - done. Odoo is connected to freepbx - done. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. Thanks

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    We’re looking for someone who has extensive knowledge of VOIP services in Odoo with OnSIP. Also good knowledge of VOIP Asterisk server and VOIP integration for Odoo. If you're an expert in this job, please apply and others are not acceptable. Please submit your proposal with your best samples. Thanks

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    Need a individual that can develop me a connection between my asterisk voip server to a crm system i build in house.....

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    I must have made the following shell script with associated cronjobs. It's fine if you send the script and the cronjob command to be inserted here. 1. script The script must activate this command every night at 01 (1 am) /usr/bin/mysql -u -p -h -P 3306 -D -e "update calendarEntries set timeStart=DATE_FORMAT(NOW(),'%Y-%m-%d 10:00:00'),timeEnd=DATE_FORMAT(NOW(),'%Y-%m-%d ...

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    install openvpn and fix the connection issue in asterisk and fix some issue

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    my provider's server crashed (raid issues) and i need a find gentleman who can install freebpx / asterisk and secure it on a cloud server having ssh access...

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    Hello Eremin P., I'm interested to talk to you about our project. We are a small group of professionals that have a few FreePBX solutions implemented in some of our customers and we are developing a web interface that has the ability to show queue statistics in a more dynamic and intuitive way and also have some agents control with the tool. Eventually we would like to implement dialers and o...

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    I have an asterisk server, the calls are droping as soon as the call reaches the IVR, I require help to fix this problem urgently please.

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    We have a Asterisk/FreePBX-system with 2000 preconfigured extensions. We need a software to run on another Linux machine that continuously monitors the status of the Asterisk/FreePBX system state and triggers email-alerts if the server is unresponsive. The software also need to log in a local SQL DB when a PJSIP-extension/endpoint on the Aserisk/FreePBX changes state. Details of the Endpoint shou...

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    We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single m...

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    our Gsm Gateway software use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

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    **To do without any commercial module** I have a requirement of Asterisk/ FreePBx to make it redundant. for that what my client is looking for is One main PBX at Location A and another PBX box at Location B. In case Location A goes down for any reason then Location B should kick in and continue the phone service for the company. Is this is something you can do? Please let me know.

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    For custom made Invoices to bill clients for incoming calls through our Asterisk PBX or for items sold and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change templates to pdf without loosing formulas and calculations (Foxit ?, etc.) 3) Linux 4) MySQL DB 5) PDF 6) PHP 7) KVM qcow2 (if possible, for disk management) We us...

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    We are looking for a Call Filter solution for Asterisk .Call Filter will Whitelist and Blacklist numbers using rules.

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    I have an asterisk server which has oem set functionality which I want to be accessible via an android app. Below are the requirements. Allow users to be able to call each other within the netowrk via UserID - users must be able to call each other at anytime i.e. constantly connected to the SIP Include speech synthesis in the app - so user can type in what they want said at the start of the ca...

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    Create a JavaScript program that will calculate the volume of the cylinder. The formula needed to calculate the cylinder volume is: V equals r squared asterisk times pi asterisk times H The height (H) and the radius (r) are arbitrary, while Pi must be defined as constant (3.14). Record the result using the alert method or document.write. Note: Send the work as an html (zip) file.

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    Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.

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    Hello, Need to setup audiocodes with asterisk, 4 analog lines and route to trunks. Work is via Anydesk or teamviewer. Thanks in advance!

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    Set up Kamailio in front of asterisk server to proxy webrtc/tls /tcp / udp to asterisk server on private network with failover. - rate limiting sip traffic by source IP - dropping malicious / invalid packets - integrated with APIBAN ([login to view URL]) - MySQL integration to be able to adjust config (like primary / secondary / tertiary failover etc) - auto install script so this can be reinstall...

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    What shapes are needed and how they need to look: strLine: This variable will hold a line of asterisks strRectangle: This variable will hold a rectangle of asterisks strTriangle: This variable will hold a triangle of asterisk. strSquare: This variable will hold a square of asterisk. shapes should be able to change orders in that they appear.

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    Hi I have CCs Zycoo coocenter-S30 , need many modification for the following : 1. Change GUI for monitor user Interface ( add counter for the queue ) also Agent user (Arabic language) 2. add some math equation in reports (Arabic language) thanks

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    We have a basic setup ready with chan_dongle and M35 modules. We are able to run this succesfully with Asterisk. At times asteriks crashed intermitantly . Some time the gateway runs for 24 hours without issues. But some time it crashes with in few hours. We need some one with experience on Asterisk and chan_dongle who can resolve the issue for us

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    We looking for a VoIP / Asterisk PBX professional that is great at sharing knowledge. We working on a VoIP solution that will be integrated in our complex public transport system. Currently we have implemented a simple 1 to 1 voip call but we planning to extend the implementation. The features we want to implement are following: - Group calls (1 to many) - Custom headers for additional configur...

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    Seeking for someone who can help with a sip integration. Please only people with experience in VoIP and / or jitsi. Thanks

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    We are a wholesale distributor that needs a way to input a list of orders and automatically call the customer and play a pre recorded message.

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