I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.
I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.
...destination and duration of these expenses (local / national / international) The detail for these reports is found in the CDR's (call data records) obtained from the customers PBX. We have a self-developed, web-based cloud platform (the back-end is on a Dot net platform,. The front-end is based on HTML, Java Script and CSS) that is providing this basic
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be
Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de
Only for gurus familiar with Bitrix who can setp Bitrix application to work with our PBX Need to use Telephony REST api
...Anton F., I noticed your profile and would like to offer you my project. We can discuss any details over chat. Below a summary.. I have a IP PBX with 2 sets of 10 numbers. Set nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503
...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....
...phones system and microtik. i had another IT Specialist setup a mikrotik router in each one of my locations. i have 2 locations. They are connected through sstp vpn. i have a 3cx pbx in location 1. location 1 has 6 phones and location 2 has 6 phones. Network access to both locations from one another are working perfectly fine. i am having trouble with phones
Hi, I have Php house made CRM, want to integrated 3CX PBX VoIP with CRM, Adding extra field in the customer profile (email) and generate report. More information are available in the attachment.
Develop a new web service, “Neverwrote”, for managing virtual notes—The backend shall run on a Node.js server, and that the frontend shall be built with React. You will only need to work in the api/ directory (for the backend API) and the frontend/ directory (for the frontend interface). The NGINX server is already correctly configured, as are all
...FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from a paid PBX (switchvox) and move to a
I am looking for a ...10 digit local number we want 9XXXXXXXXXX sent to the PBX as it will strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765
...have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [kirjaudu nähdäksesi URL:n] or yate or besip [kirjaudu nähdäksesi URL:n] up to you as
We have a company based in kuwait and we want to set a " Autodial" management system for our sales team of 5 pe...customers which is assigned by the system to each extensions call will be thrown to them as soon as each people keep there telephone down. We have following things 1.) GRANDSTREAM PBX with telephone 2.) Servers if required. 3.) VPN set up
Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem
...documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call T...
To permit clients to fill out forms and documents on our...for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already [kirjaudu nähdäksesi URL:n] is a Linux only project !
Quiero una grabación que de trámite a las llamadas entrantes con una duración máxima de segundos. Texto: "Hola, te estamos comunicando con uno de nuestros asesores. Esta llamada no tienen ningún costo y puedes hacer todas las preguntas que necesites para que tu viaje sea una realidad"
**Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development
I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?
...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message
We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information
...system 53 - Insurance lead generation 54 - Inbound Call Tracking 55 - Outbound Call Tracking 56 - Call Analytics integrate Asterisk call center with CRM to have some feature bellow: - Sale can use Call feature on CRM via Asterisk. - Can use feature Click to Call? Please show me the solutions (brief, draft with your paper) to continue!...
I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...
Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]
Atualização do site, segue as atividades| 0.1. Conversor de tabelas para Rocket (central asterisk) | 0.2 Integração entre o Rocket (central asterisk) e PROSPECTS (area administrativa) | 0.3 Inclusao do Te Ligamos (click to call) e botao blog no menu area cliente do site 1. Blog (do 0) | 2. Integração com Rd Station | [kirjaudu nähdä...
...I have 110,000 phone numbers to call bi-weekly. The problem I need to overcome is detection as to whether I am calling: 1. a basic residential type answering machine or 2. a PBX style answering machine which has 4 different ways of asking you to leave a message by pressing 0 or leaving a message for extension 101 or other combinations. The problem with
...have 110,000 phone numbers to call bi-weekly. The problem I need to overcome is detection as to whether I am calling: 1. a basic residential type answering machine or 2. a PBX style answering machine which has 4 different ways of asking you to leave a message by pressing 0 or leaving a message for extension 101 or other combinations. The problem with
...configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with other freeswitch, asterisk and fusionPBX installations. The task is to look at my setup, determine the problem, provide the solution and provide documentation of the problem and solution. Note 1. You
...integrate Asterisk server with incoming and outgoing call center for our business in US. 1. Need a person who knows exactly how to do that using asterisk and what other external integration options we will needed to integrate. 2. Hardware requirement for implementation. 3. We should be able to make call from API over webrtc using asterisk underlying
...both a client and admin portal. Users can register and fill in personal details. It connects to a MsSQL db There are links to a SMS gateway, as well as integrations with ASTERISK phone server and Stripe payment gateway The application is similar to an accounting suite, where invoices and other payments are entered. The application then runs a payroll
I need a cloud pbx program for telephone. it needs to have a menu and some sub menues, take numbers as imput and store them in a database. record audio files, play audio files, have text to speech that will play back based on rules. take caller id, and store all the information in a database. i am uploading a sample file with the program that i wrote
Need someone to install, configure and test asterisk a2billing (for calling card service use) on a Google Cloud VM. Only bidders with proof of prior/similar work will be considered. Asterisk realtime experience a plus. Thanks.
Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)
I am looking to create a SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own)
I have only a openwrt Router like TPlink Mr-3420 v2/v3/v5/ i want to run 32calls you can connect server to local a VPN and MASQUERADE connections or you can install any SIP PBX on local route like sofia-sip free-switch and register server to local if you want you can install a SIP Signal service on openwrt and pass call to voip device also you can bridge
We are looking to skillful guys to Configure 3 SPA3000 devices & 1 GSM gateway with xCALLY contact cloud center (Asterisk base call center system). The job will be mainly sitting the devices to be connected with xCALLY systems using the sitting parameter of xCALLY GUI. * the xCALLY is hosted in VPS @ germany. * The VPS @ germany is connected to NordVPN
I need an expert to give me some tutorials on how to configure my vos3000 with my buyers and sellers details along with the rates and how to secure my vos from being hack...on a daily part time basis at $11 AUD an hour. This part time employment will be on a long term basis. After your tutorials in vos3000 i would also like some tutorials in Asterisk.
i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not