Dialer asterisk activexроботи
asterisk I need to configuration client said on openwrt (open wrt running on tp link router)
We are looking for a few highly motivated professionals to operate a predictive dialer for our business. Individuals must be able to handle making calls back to back for the duration of their shift. Must have at least 1 year of experience and speak clear English. Must have own computer with high-speed internet and a work space/desk. This will be a full-time position and we are looking to open up 2-3 positions. Bonuses are possible based on results. Please send resume in a message if you are interested. Pay is dependent on experience.
I need an Android app. I would like it designed and built. It should record both dialer and the receiver audio till latest Android version 10.
Need to setup dialing script/bot. I have working Freepbx with 1 usb dongle. AUTO DIAL BOT WILL PLAY: Each time when someone pickup have call from BOT, will play 3 steps record. Step-1: Hello, we are calling... Step-2: Please choose 1 or 2... Step-3: Thank you... IMPORTANT Calling from 2 new dongles (sim cards). Can be done in different way. Each time I will upload new dialer list from XML. Can make 2 small list for each dongle. Need current status list (choice1 and choice2),after job is done. Web view or different way. BOT: Each number 2 time try, after no respond again after 3,4h. When some one will call to dongle number - need to rout call to Play 3 Step record again. Extra need button with time set to start BOT DIALING (ex. 9:00 or 12:00) VERY IMPORTANT INFO: I have working ...
We have a single channel SIP account with Skype Connect (Skype Manager) as a trial, more channels will be added. We want to integrate to our asterisk server to make outgoing calls via Skype. Please, we only interested those freelancer who have done it before and not those wanting to experiment. Project is only consider done when we can make calls from SIP phone to PSTN number via Skype Connect Thanks
I am getting error on the asterisk regarding the packet failure Below is the error [16:24, 12/9/2020] +91 98300 52277: han_sip.c:4151 retrans_pkt: Timeout on 588608927-1677186229-42297175 on non-critical invite transaction. [Dec 9 16:24:18] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 2000998286-245292724-1360354902 on non-critical invite transaction. [Dec 9 16:24:18] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 671697185-364125519-1082815926 on non-critical invite transaction. [Dec 9 16:24:20] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 832013417-1702615591-33504755 on non-critical invite transaction. [Dec 9 16:24:20] NOTICE[1975][C-0001ae39]: chan_sip.c:19537 send_check_user_failure_response: Failed to authenticate device <sip:100@>;tag=...
Adaptor needs to work in intercom system (intercom for multi-apartment buildings) Adaptor based on Ruspberry; Receiving analogous signal, and transfering it to digital; Video translation through RTSP; Adaptor registration in Automatical Telephone Station, type Asterisk, 3CX;
ClassifiedAds Web Page Final Part Modifications To Front & Back End. For the Admin Panel Of The Web Page a) The Price Label & Actual Price That is to be displayed in the listing page can be displayed or hidden using controls in the admin. b) A Mobile / Handphone Number Field To be added and this info is to be displayed in the listing… but “hyperlinked” to the phone dialer in the mobile browser. When the link is clicked it will goto the phone dialer. c) The Number of Views for each listing is kept in the admin panel… for each listing…. This info is to be displayed in the front end listing… besides the “listed on info”. There should be a Reset button to reset this number of views to “0” in the admin&helli...
Hi, We need a shell script, which runs every 5 minute, it must go into the folder (/var/spool/asterisk/voicemail/default/271283/INBOX/) and check if there are any *.wav files, if there is, then it must rename the files to , if there are more than one, then after "Besked" then it must insert a number, e.g. , VoicemailBesked1.wav... The script must hereafter attach the file in an e-mail and send it to @ The subject of the e-mail must be Ny telefonsvarerbesked fra [telefonnummer] Main mail: Du har modtaget en ny besked på din telefonsvarer fra [telefonnummer] d. [dato], [tid] til [Firma] Lyt til beskeden i vedhæftede fil. Med venlig hilsen NordicCall The above information you will get from the txt file, which is placed in the
Hi I need a freePBX ASTERISK Expert I have some little task which need to be completed as soon as possible I want to make my set up very secure I want the phone to be connected to PBX server trought VPN server Need to fix security certificate and mail SMTP for now
mettre en place un cahier des charge pour un sophtephone ou webphone j'aimerais développer un shotphone, j'ai déjà un serveur Asterisk en place qui fonction et il me manque que le softphone pour mon application
Hi, I am looking for someone who is capable of designing an android app for dialing, they must have the following features: - AES256 security during call - Caller ID change option connected to my PBX - Ensures SIP client is connected - Login Portal before dialer appears - Hide SIP info / Details This app will be used within my company for certain staffs. Let me know if someones capable.
...displayed when requested. Movement commands for the robot will be entered at the console and the floor can be displayed at any time by selecting that option. When the robot reaches an edge, it will wrap around to the opposite edge. Print all results of the robot’s movement to the console when requested. The robot can be placed anywhere on the grid when the application starts. The pen can draw an asterisk ‘*’ or the number 0 (zero). An empty grid is composed of the number symbol ‘#’. You should have options to clear, display, save all robot operations/movement to a file, and read a file containing robot operations/movements. Actions that must be saved(Case does not matter) U pen is up D pen is down <direction>:<integer> spa...
Greetings, Thank you for showing interest in our project! We are looking for tele-callers worldwide to work on our wide range of calling campaigns. Anyone can apply for this job, if your English communication is excellent, but bpo experienced candidates are preferred. Our campaigns target, 1. USA 2. UK What we Offer: We provide necessary training for the campaigns assigned. We provide data, dialer, VOIP for the work. You must be willing to work full time (9 hours a day, one hour break included) [business hours of the targeted country] Payments are made fortnight (bi-monthly basis). Pay is strictly on commission basis, no fixed monthly salary or no hourly payments or no milestones. Please bid only if you agree to work for the terms and rates mentioned above. Thank you
I need a asterisk help i have server with asterisk .I want to see the password of rejected sip user in asterisk..
Necessito de uma orçamento para criação de um chat de voz em php baseado na plataforma de telefonia asterisk
I have install Fusionpbx and set gateway (working), extensions, inbound and outbound rules, registered SPA508G phone. Inbound calls work, outbound calls and extension calls all give Temporarily Unavailable. I'm sure it's a quick fix for someone that knows Fusionpbx, I come from asterisk/freepbx, so it's all new for me. I will give teamviewer access to both linux box and Fusion website.
I have to setup a 2000 SIP trunk IVR Bulk Calling setup, Need a experienced Asterisk PHP Linux CentOS developer who have depth knowledge of asterisk and all the IVR Calling setup.
Hi Ajay J., I noticed your profile and would like to offer you my project. We can discuss any details over chat. We can discuss any details over chat. Please let me know if you are available to work on this project. just need to connect vtiger crm with asterisk (both installed on our vps)
Hi Mohammad Abu S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. Please let me know if you are available to work on this project. just need to connect vtiger crm with asterisk (both installed on our vps)
Hi, I have a crm set up and running, Vtiger self hosted on a vps. I have installed asterisk on the server too and need someone to connect and set up asterisk with Vtiger crm. (vtigerasterisk connector is installed too but need support to link it up, or use another alternative all together). This is a very urgent job to be finished!
I want to develop VoIP Traffic Analysis: SIP + RTP and web base dialer my current running server need with ui, my server is freebsd operating system and for sip traffic on opensip
we have a server pc for an asterisk call center but we need to connect the local network and provider network at the same time and also knowledge about freepbx and linux
Greetings, Thank you for showing interest in our project! We are looking for tele-callers worldwide to work on our wide range of calling campaigns. Anyone can apply for this job, if your English communication is excellent, but bpo experienced candidates are preferred. Our campaigns target, 1. USA 2. UK What we Offer: We provide necessary training for the campaigns assigned. We provide data, dialer, VOIP for the work. You must be willing to work full time (9 hours a day, one hour break included) [business hours of the targeted country] Payments are made fortnight (bi-monthly basis). Pay is strictly on commission basis, no fixed monthly salary or no hourly payments or no milestones. Please bid only if you agree to work for the terms and rates mentioned above. Thank you
We are looking for technical solutions for our Asterisk VoIP Server setup with Android as client side that uses pjsip library. VoIP feature in our application connects one user to another similar to WhatsApp calls. We have done the Asterisk server setup but we seek solutions for following: 1. When a user is calling we are unable to identify the callerId for the second user. 2. When the second user is offline we are unable to connect. 3. Require recorded voice messages that are played when another user is busy/offline etc
Looking to set up a custom skinned rebranded VICIDIAL ISO Image for call centre application . Experience is Asterisk , Vicidial , Free PBX , Linux etc preferred.
This is a hospitality and travel lease process. the business complies with an acquisition of assets to be fur...and travel lease process. the business complies with an acquisition of assets to be further rented out. We are in the search of call centers from India or the Philippines. the project shall be totally commission based and no upfront or advanced shall be considered. The project is directly from the provider. typically based on cold calling outbound, USA is preferred. Minimum 6 agents are required. dialer and VOIP shall be taken care of by the call center. data shall be provided by us. Only a serious call center shall apply. The project shall be awarded on the base of the bidding price but rather the portfolio. So, forget about the bidding just go on for the application if c...
...following guidelines: -clever/funny -photo real, not cartoony -homey/cozy feel -shows product (projects) The card “opening” should be achieved either by having an animation or utilizing scrolling in a way that delivers the punchline appropriately. The digital assets for the projects will be delivered to you and we need 2 versions of this “card”, the only difference is that one will have an asterisk in the text with a footnote that reads “gift amount in accordance with EVP reductions” or some such joke they will understand. I have attached an example of the "cozy" element we are looking for that was on a card in years past. This is time sensitive now that we are in Dec., our previous artist was a flake that put us behind schedule....
Hi! We search a Call Center or Agent for Recruiting B2B project in Germany. Differnet Projects. Only B2B Projects ! You or your Call Center have to speak good German. Project, Script, Datas etc. its all from us. If you have own Dialer, it will be good, otherwise we have Dialer. You are bidding for Hourly Price for 1 Agent. If all is fine, we can do longterm with many Agents. You will get every 7 Days Payout
Hello, We are expanding our team of agents. We currently have a version of Goautodial 3.3/Vicidial running on a server in the cloud. We also have a trunk of up to 4 CAPS, except that it is now insufficient, we need to be able...be able to add a trunk provider and modify our call script to be able to randomly launch calls with the available trunk, which would also allow us to increase our CAPS. We also have a proposal from our current provider, which is to have a second server with the dialer and a different IP, which would allow us to make calls with our provider and get 4 more CAPS. Your proposal will have to detail the work to install a second carrier and set up the calling robot, then the detail to install a second server (cluster) with the dialer to work with our curr...
i need someone who can design a web-based sip phone. which can communicate with any SIP server like an asterisk, freePBX, VOIP, SIP Server, and the browser should support firefox, explorer, and chrome. Also, am expecting a demo which is already done or ready one, so that will agree for implementation
Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Teams 365 Skills: VoIP, Asterisk PBX, Linux
To invoice clients and credit agents with none standard Invoices and Credit-Notes. See the 3 attachments for more information. --- Please note that You must be on this site to answer, because of bugs in the system.
Efetuar a configuração de um servidor asterisk 16 que tenho na nuvem (CentOS 7) com ramais SIP Polycom, integrar Asterisk com SQL local, dashboard do Asterisk mostrando usuários logados usando API.
...Dec 04, 2020 We have a project which is a Help Call system for the elderly. It's basically a fancy Voip phone running Openwrt and Asterisk on the MediaTek MT7688 CPU. We are using a WM8960 codec connected via i2S to the CPU and we are performing some basic audio functions to check the audio (record & play) using arecord and aplay. We are getting some odd results such as noise being introduced and fast garbled recording. We suspect issues with the audio driver. It is possible some of the registers on the codec are being messed up. We are looking for some direction as we need to solve the audio driver issues before we get the Voip calls moving through Asterisk. Our first task is simply to play and record audio cleanly using arecord & aplay. Any assistance/d...
Our company is looking to set up an outbound lead generation team. Looking to hire 10 representatives that have a very high level of English speaking. We are unfortunately not hiring any call centers from India,...high level of English speaking. We are unfortunately not hiring any call centers from India, or Pakistan. The project is simple, our company is giving away free smart thermostats and we are looking for a team of people that can call a list of numbers to reach the homeowners and book appointments. Qualifications: 1. Must have 10 available agents 2. Must have 1 Supervisor 3. Must have a predictive dialer 4. Must have near perfect English with minimal accent 5. Must have great communication skills 6. Must have great lead generation skills 7. Must have experience booking ap...
We have a requirement to fine tune a VICIAL configuration consisting in the following setup: * HP Proliant DL380 server with 64GB of RAM with a planned upgrade to 128 GB HDD 1.2TB SSD 800GB This hardware has installed VMware 10.7 on it and Asterisk + VICIDIAL instances are being created on it This setup should allow more than 250 simultaneous calls but currently can't handle more than 200 so we need to know where is the problem. In addition we would like someone who can do maintenance remotely via a Windows Desktop in Amazon EC2
Small office with 4 analog incoming lines and 10 extensions. Currently using UCM 6204 on which extensions are setup. Need a solution for accessing logs of: 1- Missed calls 2- Incoming calls 3- Outgoing calls 4- SMS to be sent to missed calls only Be able to access the system from local lan
DetailsProposals Project Details 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 ...for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills Required PHP JavaScript Software Architecture ...
We are looking for someone to configure our contact center and dialer on qubicles dot io the budget is higher or $100 USD You must have experience in configure dialer and contact center. We have active 800 number for USA We want to the customers to call in on the 800 number and be routed to the correct department and agents. And we want our agent to receive inbound calls and make outbound call with our 800 number. The manager has a admin panel login where all the tools are at to configure the contact center. And the agents have a separate login for dashboard and dialer. We have different departments and agents in different cities. We want to route the calls to the correct department and agents in the correct states and cities. We attached photos screenshots of the c...
We are looking for someone to configure our contact center and dialer on qubicles dot io the budget is higher or $100 USD You must have experience in configure dialer and contact center. We have active 800 number for USA We want to the customers to call in on the 800 number and be routed to the correct department and agents. And we want our agent to receive inbound calls and make outbound call with our 800 number. The manager has a admin panel login where all the tools are at to configure the contact center. And the agents have a separate login for dashboard and dialer. We have different departments and agents in different cities. We want to route the calls to the correct department and agents in the correct states and cities. We attached photos screenshots of the c...
I have some B2B data for a dialer system that has business owners informayion and business lines. I need the cellphones. Can you skip trace this data? please help
Hi! We search a Call Center or Agent for some Sellingprojects in Germany. Differnet Projects. Only B2B Projects ! You or your Call Center have to speak good German. Project, Script, Datas etc. its all from us. If you have own Dialer, it will be good, otherwise we have Dialer. You are bidding for Hourly Price for 1 Agent. If all is fine, we can do longterm with many Agents. You will get every 7 Days Payout
To invoice clients for incoming calls and credit agents for work. Please see attached file for more info.
I'm looking for someone who can develop an integration between Asterisk (v.18) and AWS Amazon Polly. Call flow: 1) Asterisk will answer the call 2) Amazon Polly will greet the caller - part of the text is fixed, a small portion is dynamic 3) That's it. There will be no interaction (voicebot). Asterisk will answer, Polly will greet and byebye. This machine is running under Debian.
Preciso de uma api que quando a ligação entrar no queue e ramal que vai atender tirar do gancho ele disparar uma agi.
I have asterisk pbx system and I am looking to update it by adding a phone application. for example, if the phone is not answered in 3 rings it starts ringing in the phone app.
I have ongoing work related to our previous project 'apk file automatic dialer'
I need some help with selling something.