I'm looking for a tech who already has completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid
We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.
I currently have two FreePBX instances, PHX-PBX and BDQ-PBX. PHX-PBX is my main PBX, and has most of my configuration (e.g. Extensions, IVR, MeetMe, etc.). On BDQ-PBX, however, I have a Sangoma A200 card and a single PSTN connection on Port 1. What I need to do is the following: 1. For all inbound calls to the PSTN line, I’d like them routed from
We want to use our freepbx as a call center solution. We need to develop an auto dialer module similar to Vicidial for our freepbx system. FreePBX is a great PBX system. Now we need to make it a complete package for telemarketing. Most of the ViciDialer should be there. Regards, ALSOFT
I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system
...Please see the attached image. The red colored box is the problematic CDR and is coming from the GSM gateway. The green colored box is the correct one that is coming from another VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in
...checkout page the fields, Pais, Endreço. Cidade,Estado, CEP, they are [kirjaudu nähdäksesi URL:n] I do not know why the red asterisk does not appear. I installed the plugin WooCommerce Checkout Field Editor, although I enter the fields as mandatory, the red asterisk does not appear , and it is not possible cancel the writing (OPTIONAL). mysite: [kirjaudu n&a...
Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for
Asterisk PBX free software programmed for six internet phones which we own. Three 800 numbers which we now have to be transferred to this system. Standard small business features: Transfer calls, Voicemail including after hours message, desktop window showing who is on a phone call, etc.
I have FreePBX installed (centos 7) need to install openvph client to connect to pfsense at ITSP (me also) pfSense. also need script to restart openvpn if ping fails. Should be a 30 minute job. Can you help ? Have access to everything pfsense and the freepbx via teamviewer.
We need a Android app that take GSM Call and SMS to Send to Server so other side phone can see SMS and take calls instantly to avoid Roaming. Also maintain call logs, SMS logs, contacts on server and sync to APP that installed on other phone. We will provide you Linux server to setup.
Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call
...We offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers;
Build a full-featured conference web app. Global Reach & Best Voice Quality. Make calls to all countries on Tier-1 carrier networks with minimum latency and clear voice quality. something like group call on Facebook or Hangout. A chat room with specific members NOT anyone has the link can enter the room!!
...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a
App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.
I have Freepbx latest version with 4 Trunks. I need a call broadcasting script. User needs to upload a wave audio file & add numbers in a text area in coma seperated format. Number of retries option, call schedule date & time option should be present. its a campaign. Calls placed, calls tried, calls answered should be visible in a page. Admin can retry
We require someone hands on experience in installing goautodial or vicidial solution on Google cloud computing. Good understanding of Linux, Asterisk and Vicidial is essential. We wish to start testing Asterisk and Vicidial in the cloud from Google. We require setup, testing and ongoing support. Phase one is setup so we can start testing. Phase
...outsource for those tasks when needed. Currently i can be specific on a project which you can tell me you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry
zoho's phonebridge does not fully support asterisk 13, probably because of java issue causing the plugin not to send the correct information to zoho i need to be able to get full feature set from the phone bridge : caller id + popup + call duration once answered click2call + call duration once answered all call data should be listed in the crm based
I am having problems getting a dhadi/Asterisk/POTS configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work properly. Config files and CLI output: [kirjaudu nähdäksesi URL:n] Dialplan: [kirjaudu nähdäksesi URL:n] The machine once finished will backup and replace my old Asterisk servers that basically operate as POTS
I have setup a Trixbox CE server, now i want receive call that r forward thru ip to my server Just a small project, i have more work on the server later : *Custom scripts, for example, call from a list (excel or txt) automatic *Maintain server, Daily, weekly, add new trunks & routes *Security
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 Transfer • Blind Transfer • Supervised Transfer...
Project ICT VoIP – Real-Time VoIP Billing for Current WHMCS version ICT VoIP Billing Panel addition (existing) VoIP Extended Rates Package Rates Extended New* - Enable/Disable toggle for Real-time Billing (just below VoIP Product/Package) (update table) New CDR table to collect all CDRs from [kirjaudu nähdäksesi URL:n] CRON All fields are ...
...backups. I also need to set it up with a Control panel like Plesk or cPanel. It needs to be setup with a hosting server, mail server, SSL for domains, and I will set up an Asterisk server in one VPS which also needs to be secured. I also need help migrating over from my current VPS to the new dedicated server setup. If you think you can do this please
Hello Daer we are looking for an account managers who did already work in voip field , which has the ability to discuss prices and negociate offers with sellers ,
Hello, i want script to Test sip accounts with Back SIP response codes [kirjaudu nähdäksesi URL:n] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent
Looking for VOIP app to be created for IDD calling, SMS, free call and video call apps to apps, balance transfer from app to app, top-up recharge The Virtual number will be the subscriber’s MSISDN Someone who has done VOIP app only contact. Don't waste time if haven't done it. I dont want proposals saying we can do it we have team...etc
Pjsip c/c++ Voip doorphone echo cancelation and noise cancelation example I develop doorphone application with embedded Linux using pjsip but I have problem with Ambient noise and with echo. I need solve
Voip app Qt/QML PJSip c/c++ example for Android /iOS and Windows. I need too example of echo cancelation for use in doorphone e with pjsip
VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist
We need a project manager who can describe the needs of building the app, and make the app work. The design have to be connected to a Kolmisoft Mor switch
~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution...new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).