Opensips openser asteriskроботи
hello sir i have a projet of voip in morocco but this period i have a problem in my system based on asterisk+A2belling ;i must do a filter to block any number enter more 12 digits and less numbers called number robotic or test number
We are looking for an asterisk / full stack developer who can help us build a call center management platform platform and dialer using Asterisk. Here is the basic outline of the project. This project will require the following. Agent Control (Manage agent and customer interactions) Call Routing Queue monitoring and reporting Provide call information (referred to as call pop-up) to agents before handing over the call 500 outbound channels sip (spanability) administrator gui phone list builder dnc database and scrubbing tool statistics page wav file uploader and player for outbound email for call results .wav file data distributing leads based on availability of agents to multiple customers the completed project must be distributable. If you have built something similar, pl...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or ...
Hello! I am looking for an Asterisk (VoIP + SMS) developer. I have a Dinstar Device with 8 GSM SIM Cards that can accept and make GSM calls. I need those calls forwarded via VoIP (preferably Asterisks) and distributed equally among Call Center Agents that are currently logged in. This device has a capability of Sending and Receiving SMS, and we need that feature too. A software was already made for me, and it is working great and is hosted in Digital Ocean! However, I need more features, such as reports, scalability, long SMS thread, among others. However, I can no longer contact the developer. He is no longer logging in to his Skype for more than a year! Thank you! ~mark
I build Vtiger server with multiple databases for each organization. The connection to the relevant database set with the login parameters. In addition, there are multiple asterisk servers each is multi-tenant providing services to multiple organizations. I need to connect the Vtiger to the asterisk (the asterisk server ip set in the vtiger relevant DB parameter). The asterisk AMI connection is as standard asterisk server except each extension as his relevant organization as well. In addition, there is a node.js server installed that already connected to all servers and extract all events.
I am looking for a detailed instruction for the installation of Opensips, Opensips-cp, freeswitch, homer, cdr/billing(eg cgrates with gui), all in a single machine (real or VM). The host system would be Debian 10 64bit. The documentation should be taken from a novice engineer and get reproduced in a step by step manner. Everything would be in LAN and the IP would be with gateway/dns 192.168.178.1. Configuration of the users should be described how to get started. Describe how IVR, Voicemail, faxing is enabled. Description of how to create cost plans(pre-paid and post paid) for different extensions and how to extract CDRs and billing info for each extension. More jobs to come!
I NEED A SOLUTION TO FIX THIS PROBLEM. " no RBT, Fas voice recording with charge "
I need assistance in setting up my FreePBX home server. I'm using Cisco WebEx Teams and want to use the call in feature using the SIP URI provided by Cisco (example sip URI: <extension>@). So I've set up an extension with the DIAL property pointing to that specific SIP URI (which means that I'm not dialing the SIP URI from an internal endpoint). The problem is that Asterisk is using UDP and looking for _sip._udp. SRV records but WebEx is only providing TLS transport. Now I don't know how I should set this up in order to force TLS for this specific external SiP URI.
I am using call center on Asterisk there are library with and When hold and resume, can not process this event. Web phone built with angularjs Please help me.
I am an experienced PHP coder. I have built 2 major projects using Twilio voice and sms features. I have another client that would like a phone system created. I would like to explore my options with Asterisk. I would like help and guidance from someone with a lot of experience in this field to help me choose the best option and set up the proper platform. If this goes well, then it may be worth it to migrate my current twilio projects to Asterisk or another platform as well. I am looking for someone who speaks English as a first language and someone who can be available working hours of EST.
Buenos días, Queremos integrar Asterisk 13 (con freepbx 14 y por debajo Centos 7) con ZOHO CRM. Los usuarios trabajarían desde PCs con windows y softphone instalados. Muchas gracias saludos
I use a production voip server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and...server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and send us caller details via growl desktop pop-up. this is very easy for php developer After last few system modules updates, this is not working properly (we have only caller details, nor called line info), any php developer can fix this, no need to have voip or asterisk skils. plz contact me for additional info (here : desktop growl popup notifications look - 2 files which have been edited last time - budget fixed : 80 $ in one day - bid only if you re interested and ready to start withi...
Hi, I'm buying existing code or script in various theme. It has to be fast, modern, bug free. Anything can be interesting but the following preferred: Saas, CRM, Real estate, video chat, booking system, asterisk, webrtc, etc. CODE MUST BE ALREADY DEVELOPED AND YOU HAVE TO OWN THE CODE (!!) and connected website has to be presentable with demo back end. I do not care codecanyon links or "I can do it for you OR I'm good in x or y program language". Thank you for understanding - I have very bad experience with auto bids and fake promises. Cheers
We are trying to fix the one way audio issue on Freepbx
hello, i have working asterisk fresh installed server. i need to add my voip trunk into it + my dailplan need Voip expert on urgent complete this configuration
i would like to create a dial plan and the workflow will be as followed there is 1 DID when this did is dialled it will connect to an asterisk server... this will ask for the caller to enter there pin number .. if the pin number exists it will then ask for them to set a caller id ... then it will ask them to enter there destination number which they will be bridged to
...detail is an absolute must. Issues must absolutely be resolved within a give SLA's timeline and constraints, follow ups, KPI's, resolution rates, customer satisfaction and other factors do play a significant role. You will be working with Exchange, Active Directory, Windows 7, 8, 10, Windows Server (virtual and physical(HP, Dell, Lenovo)), Linux (Debian & CentOS), MS-SQL, PostgreSQL, MySQL, PBX (Asterisk, Panasonic, Fortivoice, Mitel, Northstar, FreePBX, Elastics), Firewalls & Switches (Fortigate, Cisco, Checkpoint, Sonicwall). Networking is a must as well, switching and routing (Cisco, Forti, Netgear). As menitoned previously this is a position/contract that will incorporate multiple individuals and when we've mentioned that attention to detail is a mus...
We are looking for a guy who can create SIP based GSM gateway for us by using GSM 800 and QUECTEL M66 module so that we can connect it though asterisk application.
...asterisk16 installed via packages. I want a MINIMAL working asterisk SIP configuration to accomplish these things: 1. Three softphone extensions that can call each other. Two on desktops behind internet firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself. 2. A publicly available extension, like me@, that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me). 3. Everything secure: No unauthorized access. 4. Be able to switch between RTP and SRTP by changing the configuration. 5. If possible: no asterisk database connection, only flat files. If not possible, explain why. 6. Asterisk comes up and runs as cleanly as ...
We're looking for an experienced translator to collect all the website text and translate it from italian to english. This project requires experienc...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...
we have asterisk 11(freepbx) setup and running. need vtiger install and connect to asterisk.
We are in a requirement of sending SMSBlast/Campaign by using the SIP account. We have VICIDIAL and ASTERISK EPBX. So, by using any of these SIP accounts need to send SMS without gateway.
This project is to setup a FlowRoute SIP Trunk for inbound/outbound calling and one user (for a softphone) on a freshly installed Asterisk based server (XiVO).
Expert needed to install and configure Goautodial Call center software. the platform is based on linux centos 7. Used technologies : WebRTC Asterisk Voip
Hi, I have a asterisk PBX and my "programmer" really just did a very poor job. Basically, we moved the system from 1 server to another and again when he transfered it he did a very bad job of it. 1. The system needs a2billing added (it was on the previous one) 2. The system needs to be updated tp php7 (previous version on old server was php 5) 3. The system needs all the - customized scripts-- to work with php 7 on the new server The domain name needs to pint to a specific index file (in a folder on the server) The same file the IP address points to. He renamed a few folders also (where both the previous folder and his -- renamed edited-- ) versions are on the site. This way you can see what it - should have been-- and what he has done that does not...
Installation and customization of Asterisk 1) Install asterisk on server 2) Customize Asterisk to work as per below logic --> Asterisk should be able to receive inbound calls on DID number for different providers --> When there is inbound call on asterisk , I want system to hang up inbound call and automatically dial out/Call back on caller id from which inbound call was received. --> Once asterisk dial out then it should be capable of identifying whether call was answer by Answering machine if it is answered by answering machine then system should hang up call without forwarding to agent extension , Only calls that are answer by Humans should be connected on agent extension. In simple words i need Asterisk with Callback Featu...
We need to configure a Cisco VG224 to work in ISSABEL with all 24 analog ports. Configure dial plans and to work with more than one Cisoco VG224 in the same ISSABEL
Hi, I have a working freepbx installation. That server is working since two years without any issue. The version of the installation is 14.0.13.12. Now I want to use same freepbx server for message communication. I use X-Lite as VoIP client. I want to use X-Lite for message and file transfer as well. All extensions connected to the server could communicate each other via messages using X-Lite. Regards, ALSOFT
Need an experienced tech for Vicidial support in real time provided over Skype and email. Must be available at any time 9am-5pm PST and respond within 5min during these hours. Work load is minimal and most days does not take more than 10-30 minutes combined but must be available during these hours without exception. Must be experienced with Vicidial. Must have good conversational English. Please offer monthly price.
I need A2Billing/Freepbx Software installed on my server and configured for a soft-phone, which is a third party app on my server, the objective is that I want to:- 1. I just need help setting up dial-plan to call long distance in USA/Canada and all International and Local calls. 1-a. Dial remotely into the server, 1-b. Enter a pin number, once the pin number is entered once, thereafter, set-up a caller ID authentication on subsequent calls ! 1-c. Enter the destination that I want to call 1-d. Then the soft-phone would initiate the call. 1-e. Call history 2. Configure a method to discourage Caller ID spoofing and to activate/deactivate account - prohibit Spam Robocalls 3. Enable Search feature for admin to search by email, Account Number and caller ID, Pin Number and Account Number 4. Cre...
We are MVNO project. Asterisk Deployed. Need to connect Kamalio and SS7
I need to set a signaling gateway (like kamailio/drachtio etc) that will be use as a proxy between Asterisk server and web clients. 1. Users will use webdialer based on webrtc that will connect to the proxy using username and password. The proxy will route the signaling+RTP to specific asterisk server as a standard SIP client. The webdialer will be able to initiate calls and receive calls. The webdialer will have in addition "hold/transfer/conference" buttons. 2. CallMe button that will use a subset of the webdialer. This button should be generate a token that contain the domain, username and password. This button when clicked will connect to the proxy and the proxy will dial the preconfigured number. I'm expecting to get the server installation process and c...
Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future:...
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
We are looking for an asterisk developer to help scope and implement a project
Due to hardware problems our very old Asterisk server died on us about 9 months ago, but due circumstances we haven't been able to replace it. We are a small company with just a few phones and some other linked equipment so we made do using a direct line, our mobiles and a professional skype account. However we would like to get the situation fixed before the end of the year if possible. Due to the situation the Asterisk will need to be set up on a fairly new server (2x XEON E5-2620v3 6 Core 2.4GHz/32GB(2x16GB)) but with some old(er) peripherals like a DrayTec Vigor 3300V switch which holds the FXO and FXS connections, some grandstream SIP phones and a few regular PSTN phones, as well as Softphones on our laptops and Android phones. Also there is a SIP door intercom we ...
Summary: Our current application is housed in Wordpress. The main applications backend is connected to an Asterisk system. We would like to switch over to a Bootstrap framework and update the template used for our website. Deliverables: - Transfer the Asterisk-based application that's currently housed in WordPress to a Bootstrap administrative template (we'll provide the template with the backend admin layout) with the same read & write capabilities that are housed in our current Wordpress setup. This includes: Read & Write access to subscriber information, ability to change plans for subscribers, etc. - Update the current website to a new bootstrap template. Seeking an experienced developer, that has experience with Bootstrap, Wordpress, HTML, CSS, &a...
Software to make Templates of all kinds for Invoices, etc. To be used with Magnus/Free Side Billing. Templates are made with LibreOffice on Ubuntu OS. PBX is Asterisk 16.6. Work is to be done with Teamviewer on our servers and good knowledge of the new LibreOffice, Linux and Asterisk is a must ! -- Everything is documented and the project will be divided into 3 milestones.
Hi, I am looking for a software developer who can build real-time speech to speech language translation software. Need this done using Asterisk or FREEPBX
I need a graphical interface for the asterisk to do the following: Using raspbx, sql, php We use 10+ 3g modems (with the possibility to see their status and edit the imei from the graphical interface) We have a database with 6,000+ phone numbers ( imported in sql) We have a set of questions that we want to ask our clients automatically. An ivr-dialplan must be created, all audio messages must be TTS (text to speach and retrieved from the graphical interface, we must have the option to change questions over time.) We need to see in real time in the graphical interface the calls that are in progress, statistics, the date when it was called and the keys pressed. Appeals to the number xxx -> 1) xxx answers, we save the action in sql, we return the message: Hello, my name is x...
Diseñar portal web sobre asterisk que incluya opción registrarse, login (ingresar), DID para seleccionar o escoger (activar).
Looking for intelligent, engineer minded, polite, honest, diligent full stack developer (team). Proper English is essential. Must: all php framework, database, ftp, email, push, crone Advantage: unix server, upgrade, ssh, htaccess, centos, cwp, admin, mongo, asterisk, webrtc experience, python, rss Continuously work, bid not important. Auto-bid will be auto-refused. Please send a proper intro.
We have a log file where we have a number of ID numbers for the shell script to retrieve the time in the log as below cat /var/log/asterisk/full | grep C-0000335e | grep 'Spawn extension' | grep default [2019-11-28 21:52:48] VERBOSE[10832][C-0000335e] pbx.c: Spawn extension (default, Queue-206611, 2) exited non-zero on 'SIP/fcplatform-00004e64' cat /var/log/asterisk/full | grep C-0000335e | grep queue_time= [2019-11-28 21:52:43] VERBOSE[10832][C-0000335e] res_agi.c: : Query is UPDATE queue_calls set queue_time='13',agent='l/203@context-out/n',status='0',call_time=Now() where unique_id='1574974337.190819' and queue_name='206611' The script should then state where each ID number we have in a file, how lon...
Software to make Templates of all kinds and Invoices with Magnus/Free Side Billing. Templates are made with LibreOffice. OS is Ubuntu. You must use Teamviewer to access our server and knowledge of LibreOffice, Linux and Asterisk is a must ! -- Will be divided into 3 milestones. Everything is documented.
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 Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document and see attached summary for what is done so far
i have server cloud and IP telephone i need install Asterisk free pbx to connect our getaway
A php script to manually add a phone number and Queue ID into FreePBX's Queue call back option to be called back at front of Queue. Run as a pipe from bash. The idea is for the FreePBX callback module to do all the work we just need to add the phone/queue number into their system. FreePBX Asterisk 16.6.2
I need help for asterisk goip long term , I have special dial plans that help to avoid sim blocking , I need to fine tune these dial plans .My budget is 50$ .
