Asterisk voicexml jobs
...include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) all using: VERSION: 2.14-715a BUILD: 190705-1012 © 2019 ViciDial Group To describe the issue as best as possible: inbound/outbound blended calling is seamless with 40 users registered and taking calls while the asterisk is ra...
...include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) all using: VERSION: 2.14-715a BUILD: 190705-1012 © 2019 ViciDial Group To describe the issue as best as possible: inbound/outbound blended calling is seamless with 40 users registered and taking calls while the asterisk...
I need to place bulk calls using asterisk-java. I am using asterisk and AMI as of now. Please contact.
We need to install a WebRTC softphone por our callcenter system.
I need an app for Android. I already have the design, I just need you to believe it. always on top. h264. peer to peer lan/wifi, reguster, phone book, customs logo, keypad for dtmf dial on call, mute, history, sos, phone, video WiFi, ethernet, 3G, 4G. and push for service with asterisk, phone book, blf list.
...processing, using either python or PHPAGI. 1. Caller press 6 to enter Credit card Processing 2. Caller is asked to select a pre defined package pressing 1, 2, 3, 4, or 5. 3. Caller is asked for their credit card number 4. Caller is asked for their expiry date 5. Caller is asked to enter their ccv code 6. Read what the caller is about to order and then state the amount. Using Westpac Payway API Asterisk/Freepbx needs to hand this across to agi to process credit card in real time on successful transaction (would prefer hold music whilst the system gets approval). On success Approval - hands back to IVR system and generates a 6 digit pin number that is stored in SQL table. On failure - Advise caller to contact credit card provider. This is a fixed price contract in Australian...
We are new to VOIP and we are wanting to trial out an IP/PBX system on our server. We have been avised to consider installing Asterisk and also create a web interface to manage the configuration of each user. We require someone to be able to build us a complete stand alone system so that we can configure and test making calls in the UK.
Hello, we want to integrate speech recognition into asterisk (grandstream 6510), the callers will talk portuguese from Brasil and some specific routines are needed: issue billings based on social number
I need a development expert on asterisk, to help me and guide me to find and modify parts of asterisk code
Hi, I need an Asterisk Expert to setup asterisk server. More details through chat.
We need a simple VB.Net project that uses the Asterisk ARI to dial phones/extensions. 1 form a text box to enter the number to call, a text box for the extension to call from a button to dial and a button to hang up said call.
We four Vicidial and we want to enable Firewall on that as currently we are seeing lot of login attempts via SSH and Asterisk , so need to setup security for our server where only allow IP login .
asterisk based quiz ivr How it works? The IVR solution will randomly ask different questions to the caller, English and Hindi Caller can answer using either dialpad For each answer caller will be guided to the next question until the end. Based on the collected answers, system will take predefined action 10 Question, Four choice answer (1,2,3,4), On correct answer user gets point, On both correct/wrong answer next question is played, Questions can repeat, Phone number with points are recorded, Ranks will be based on correct answer and fasted answer, Black list number from system, Points menu etc The Quiz and Competition Interactive Voice Response (IVR) Solution can be used to ask a series of questions to the caller for different purposes. This can be the MCQs or objective que...
At the moment, we have two issues regarding the Auto Pause config and Transfer Issue on a dynamic queue. 1) Auto Pause Issue: We already setting all autopaused config on asterisk like: wrapuptime=10 autopauseunavail=yes autopause=all autopausebusy=yes But none of them is working. This caused an agent who is busy/talking to another client, got distracted with another incoming call. When do command asterisk -r queue show, the agent status is in use, but keep receiving another incoming call. We already trying to pause it through AMI, but sometimes works some times not works. 2) Transfer call issue: When transferring a call to another agent, after the second agent answers the transfer, both agent status become "Idle" (Not In Use). The first agent already hangup th...
We have a PBX running on Asterisk and we have around 60 extensions configured. We use this in order to do remote voice alerts in softphones running on servers and some Axis IP speakers using SIP. We have many Axis speakers working and for this we had forwarded the necessary ports for them to work and we use the public IP address. All of the extensions are outside of the network our server is running. For the softphone we use Jitsi and we have preamplified speakers connected in the speaker plug of our sound card. So when from our office we dial an extension, we have Jitsi auto answer the call and then we speak and call talk and the sound is heard. When we use Jitsi, we use a VPN connexion (OpenVPN) so we use the VPN (local) ip address. But when we work with the Axis IP C3003-E ...
We are looking to build an Asterisk based cloud phone system to support upwards of 5000 Users. We have been a reseller of cloud based Phone Services for 15 years, and want to build our own platform
I am providing call termination services for my clients but I am Not getting consistent answer supervision from the terminating gateways. I am looking fir an asterisk solution to solve the false answer issues by having the gateways register to the asterisk server and asterisk will determine if the call is truly answered and starts the billing. I am open to any method of determining how Asterisk designates am answer ( sop messages, voice detection etc) Please let me know the solution details, version requirements, hardware requirements, project time for completion et
I have asterisk working with goip, I need to enable sip errors in asterisk and create a script to check sims are not budget is 25$ for this project, please do not bid high amounts I will not entertain such bid if you are expert of asterisk and know goip.
...PBX platform. Required Skillset This project will require a VoIP engineer with working knowledge of Fusionpbx (asterisk) and DSipRouter (kamailio) SIP servers, both open source software. Current Environment Two servers are generally operational and hosted in AWS. Both servers are publically accessible over standard and necessary ports, 443, 80, 5060, VoIP media ports and credentials will be provided for both machines. 1. DSipRouter (SBC). This is a distribution of the industry known Kamalaio SIP server. For more information on DSIPRouter, visit: 2. FusionPBX (PBX). Built on Asterisk, this also a highly recognized VoIP PBX. For specific information on the distro, visit:
Hi Ali i need do option my the new system 1:- i need do system for the recharge card 2:- i need to do option for the ussd wich allow the employ to control all thing with easy 3:- i need add new culomn for the page which count all hit come from asterisk 4: - i need to do timer for every user which make me to control how many hour the user will used the service 5:- i need add option for the user page which allow us to stop or enable every user by check box 6:- i need add option which can control the time between the user like the max of the secund
Hi Ali i need do option my the new system 1:- i need do system for the recharge card 2:- i need to do option for the ussd wich allow the employ to control all thing with easy 3:- i need add new culomn for the page which count all hit come from asterisk 4: - i need to do timer for every user which make me to control how many hour the user will used the service 5:- i need add option for the user page which allow us to stop or enable every user by check box 6:- i need add option which can control the time between the user like the max of the secund
Hi Engineer, we like to invite members of the freelance to build a cloud phone platform with us. Engineer must have deep understanding on database, onboarding UI and linking of UI with asterisk system.
Hey everyone, I've been digging in to Asterisk for a while but I'm not a developer so I may not have all the concept right. I'm open to any suggestions that come from a professional developer. I'm looking to setup a multi-tenant asterisk server working on the Asterisk Realtime Architecture. Since its going to be used for multiple customers it must have a structure that is going to allow me to update it(as adding and editing extensions, IVRs, trunks or dial plans) real time without any downtime (not even seconds). I need to record calls by adding them to a DB so that an Administrator can listen to these records by a web-based gui in the future. I also need to voice mail service to be active. I want extensions to be able to listen to their private...
I have asterisk working with goip, I need to enable sip errors in asterisk and create a script to check sims are not budget is 25$ for this project, please do not bid high amounts I will not entertain such bid if you are expert of asterisk and know goip.
Must know, php, mysql, sms technology, voip, freeswitch, asterisk. ubuntu, python We are looking to retain a working for many projects. Our intent is to provide projects over years to come. You will have to log into our site and understand what is currently there in order to be able to edit, add , update.... Please inbox me for more info
Hi , I need a Freepbx and asterisk spec to help me on some stuffs . I will take a test with a simple task then we go through main project.
Hello, We need help to setup Analog Voip Gateway with Asterisk, model: ~ Remote session is provided with TMATE, so be ready for it. Thanks in advance
Backend software project for professionals, coded on for 15 years locally in South Africa, provides cloud file, SMS, fax, email client, cloud PBX, accounting system, and much more, need expert developers who can contribute significantly and at speed. Hurdles with old code, old technologies and many programmers who thought they can code who 'coded'. If you are 1. not an expert who is prepared to take on a challenge 2. not available 3. cannot code and read any php code at speed (no matter how bad the previous programmer was) 4. has the word 'impossible' in your vocabulary, please do not apply. We need someone who can read and rewrite existing code (often very old) in powerful ways. Prepared to negotiate substantial incentives or rates if coder successful on parts assigned...
asteriks management with phpagi libraries and outbound IVR ascertainable experience
We have a project within Asterisk IVR about 1000 lines of code containing: - Option selects - Webhook calls - Database lookups - Automated dial-out we need to change our telephony engine to Twilio and adjust automated dial out to use Twilio as well. Language considerations: PHP or Node.js Attached is the IVR script
Install vtiger connector to running Asterisk 11 and vtiger 7.2
This is to change a small software program into an ISO file for easy downloading by agents of a call center and make some adjustments to the existing software . Must use Teamviewer to connect to our Multi Disk Server.
We had to reinstall an asterisk server, and now the dahdi T1 cards are not refelcting that they are bing programmed for a T1. This work needs to be able to worked on now
i try to connect with asterisk throw pami but got some errors any help
Searching for somebody who can find out issues with asterisk and fix it. Thanks!
...and 1 day to deliver 2nd part please bid according to your time cost to complete the project. We will create initially milestone for the 1st part of the project. Once its concluded successfully we will create the 2nd milestone. 1st) Build, install and configure for simple tests the open source project Its a 1-2 hour task if your have experience with SIP and asterisk - Milestone Max is $100 and release will be on installation completion - time to deliver 1 day. Awarded freelancer should set this up on his server and we'll make a call to/from the telegram SIP gateway and if it works we will offer remote access to our server to do the installation on our server and instructions sent on how to install and configure. 2nd) Create a service like
Need configuration of the FreePBX to send SMS by gsm modem. My Freepbx works on raspberry pi 4 B with 16gb card, and has 1 modem connected with a sim card. There is 1 extension SIP trunk connected GSM to VOIP, and VOIP to GSM. Im connected to freepbx by zoiper app in my mobile phone. In short way: In my mobile phone im using "zoiper app" which is connected by SIP to Freepbx. I...send and receive sms from gsm sim card (modem is huawei e160 in raspberry with freepbx) using a zoiper or other voip app. Curently text mode in zopier doesn't allow me to send or receive gsm sms. [check images] All recived sms messages going to previously email address (spam inbox) in freepbx settings. IMPORTANT: Need skilled person to do the job, only serious performers and freelancers in a...
ASTERISK EXPERT to set up PBX We are looking for an Asterisk Expert to set up PBX. Interested freelancers can bid for the project.
Hi guys! I need an experimented guy with a heavy knowledge of Asterisk , SIP Protocol , PATTON E1 gateway, Issabel IPBX and RAD ASMI-52 modem for assistance on E1 trunking configuration between the PATTON gateway and my Issabel IPBX and E1 signalisation troubleshooting between my Patton and my ISP RAD ASMI-52 modem. Patton model is SN4171/1E15VHP.
Searching for an expert who can audit asterisk server and fix existing issues.
Hello, We are looking for the expert who can work on the Asterisk server and telephony integration.
We need help to install a Kamailio infrastructure with Asterisk, for few millions of calls per day. Solid experience required on Asterisk and Kamailio. Special Development Module for AS* and Kamailio. Development on the core code for AS*. Training. Database configuration. Monitoring script with Zabbix. GUI interface. Physical appointement in France (Paris-Toulouse) or Swiss (Geneva).
I have cdr from asterisk ( my sql db) I need to choose several columns ( filter them accurding to the info inside) Than to send excel sheet on spesific time to my email. I can add many reports as i want. It should be a tool for windows.
Create a service like We need: 1) Our clients calls from Telegram app to our Telegram account should go to our Asterisk PBX via SIP 2) Be able to from our Asterisk PBX call our clients Telegram accounts. 3) Handle 50 concurrent calls Open source example:P roject Here you can see how the bot works Libraries used: Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected to our PBX for tests 50% milestone on delivery of source
Want to convert avaya 1608-l to sip so that it can be used on asterisk pbx
Create a service like option to use their open source project so that our clients can call our asterisk PBX from their telegram accounts. Here you can see how the bot works Libraries used: Supported call types are: SIP->TG call by username SIP->TG call by phone number (if callee is registered in telegram) TG->SIP to fixed SIP URI (can be set up in TG2SIP settings). Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected
Can anyone make avaya 1608I work on Asterisk pbx
Searching for an expert who can audit asterisk configuration and can make the PBX secure from outside.
