Asterisk chan dongle jobs
Hi there, Hi I'm looking for someone to assist one of my clients on site in Pyrmont NSW (sydney) - it would involve fixing issues with HDMI cables and finding out why people struggle to connect to the TV - also setting up cameras etc. The important thing is to spend the time finding out why people have had issues with this and documenting it and explaining it so it's easy for them all. Also setting up the video conference system. I work remotely from Germany so it's difficult for me.
Need someone with experience in answering machine detection. All we need you to do is modify the and allow our agents to test that we are not getting voicemails. I understand that it is not possible to get 100% detection, however the amount of voicemails we are receiving is not what it should be.
We are looking for a virtual phone solution to be accessed with iPhone that can have an auto...iPhone that can have an auto attendant (option 1: Sales and 2: Support). I need to build our own solution, especially we don't receive many calls. I expect that the solution would be in the cloud. The system will have a US number. The developer may utilize technologies and system components like soft PBX, SIP, Asterisk, DID, etc. I need a stable system that would have a cheap running cost. Please give me a plan of what you will do and a fixed rate. Example of optional technologies: Asterisk, FreeSwitch, Kazoo VoIP Cluster, Kamailio, callweaver (faxing), ASTPP, Auto Dialer, opensips, kamailio, Call Center, Elastix, Vicidial, VoIP, Ringless VoiceMail, 3CX, FusionPBX, Issabel, OS...
I have flexisip , Asterisk , Opensips working well . I want experienced people flexisip , Asterisk , Opensips Who can support configuration document. Client <--->Flexisip ( Frontend )<---> Asterisk ( Backend )<---->PSTN Please check this link and If You can configure and support contack to me.
I am running a FreePBX instance of asterisk and I was hacked, lots of outbound calls, I think I blacklisted the IP that hijacked an extension and made the calls. Now I see a lot of attempt in my CDR Report with "Congestion" under app and s[from-sip-external] under destination. System number keeps changing and the CallerIDs are all 4 digits. Need someone reliable that I can use for this PBX and other Asterisk enhancements and improvements.
need script to change IMEI for USB Huawei dongle on raspberry Pi ruining on centos 7 minimal
I have flexisip and Asterisk , Opensips working well. I want experienced people Who can support configuration document. Flexisip () Frontend Asterisk , Opensips Backend
We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outbound Calling. Caller ID must be passed on incoming calls. Ability to have more than one Whats App number to work simultaneously
hi, i'm looking for a developer on asterisk, to create predictive dialer with open screen operator callcenter
How much money take you for install asterisk for link with app for retails calls
Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work
Goal ------------------------------------------------ Tha goal of this project is to enable our operators to issue an SMS message to a potential customer linking to Google maps with instructions on how to get to our location from their current whereabouts. Background ------------------------------------------------ we got a lot of callers looking for information about directions. The address alone isn't much useful getting them to the location, simply because we live in a very rural area. We've come up with the idea to be able to send them a link to Google maps directly to the phone from the phone call itself. This way we are hoping that the other was received the link click on it, Confirm they know where they are going, and we won't have to deal with repeat calls later o...
Looking for an expert to integrate Vtiger with Asterisk FreePBX What Exactly I need is: 1.Click-to-call right from a lead or contact record to save time. 2. See the callers' contact information on the screen whenever they call 3. Quickly create opportunities and contacts right from the incoming call popup 4. All calls are logged so that you can refer back to call histories later 5. Automatically record calls and link them to contact records in case you missed a detail
Hello, I need some help in Asterisk IAX2 trunking and callflows over it. Experience in the fork Elastix or Issabel is a great plus. Will be a half day job or a bit longer. Kind regards, Danny
I want to develop a application that can do real time call transcribe in a call center environment, you must provide your description of solution in detail on how you can push the audio stream from a traditional call center architecture (IVR/ACD) or in open source environment like Asterisk. how to classify which audio stream belongs to which agent. Currently we already developed a demo code which can receive audio stream from Twilio through ngrok but we understand that in real call center environment it is more complicated which involves multiple agents working at the same time. we study the IBM solution in voice gateway and understand that SBC (session border contoller) probably needed to fork the call and audio stream to voice gateway and further to Rest Server. we need an e...
Need a caller Id spoofed prefrably done on asterisk it any platform you see fit
I’m need a caller Id spoofing system done on asterisk it any platform you see fit with easy interface prefably connected via mobile
I need a system on asterisk or best platform to make spoofed calls need simple interface and preferably to connect via mobile device
I need a system developed on asterisk for caller I’d spoofing where I can input my own number si caller sees this I also have another ivr project needed after depending on success of this project
Hi Alex, I am looking for an Asterisk expert who can help me a bit out with IAX2 trunking and callflows over it. I need a half day of education. Kind regards, Danny
I have an issue with Asterisk FreePBX configuration. More details in chat.
Hi Chan P., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
Need to add a Number and Voice mail to and exisiting Asterisk FreePBX server.
1) Install vTiger CRM via Installatron (cPanel - GoDaddy) and manage inbound / outbound calls. 2) No sync between IPPBX (Asterisk) and Inbound calls (Exotel). Currently customers dial a vanity number and connect to respective agent groups via Exotel. Agents see only a PRI number whenever a call lands on their mobile phones. Outgoing to the same customer is made via GSM gateway connected to a IPPBX. In order to allow my agents to handle calls / follow-up with calls, we would like to receive calls on our own PBX using the GSM gateway and manage the calls via vTiger CRM integrated with ERP for database.
I need a java application that can upload a FreeSwitch or Asterisk CDR to a MySQL database. Identifying every PBX with an unique ID. I need get the code and not only the .jar.
I need you to build it. i need asterisk voip server setup done on cloud server
...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
Scope of work We’d like to create a Proof of concept solution that allows us to send some data (images) using analog fax / phone line. The solution consists of Raspberry PI device and usb analog modem dongle. User is able to send images using http post request to the http server launched on the device What we want - Raspberry PI setup connected to the analog fax modem that can send provided image to another user (phone number) - User should be able to connect to the device and send the file using provided script e.g. 55512345 - Once the basic setup works we want to be able to upload photos remotely via http server launched on the device What you need - Raspberry PI 3/4+ - Analog usb modem (e.g. U.S. Robotics) - Analog phone line - Access to virtual fax service, or
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
