We are currently seeking a skilled developer to customize LinPhone ([url removed, login to view] and call statistics per user, daily, weekly, monthly, etc. Manage users, receive their complaints and assist them. This job requires good skill in App development and SIP (VoIP). If that is a work you can handle, please bid.
Dear Sirs i would like when someone calls, the sip phones to show up his name that i will have previously stored in the freepbx phonebook page i tried in the "Caller ID Superfecta" page of freepbx to enable the "Asterisk Phonebook" and in "CallerID Lookup Sources" to select source type:"internal" but nothing changed Your help please
I am looking for a freelancer to help me with my project. The skills required are Asterisk PBX, Cisco, FreeSwitch and VoIP. I am happy to pay a fixed priced and my budget is $250 - $750 USD. I have not provided a detailed description and have not uploaded any files.
...configure custom SIP gateway in our FusionPBX for the outbound calling service. Currently our SIP provider unable to provide us direct SIP settings to configure them with our FusionPBX as they only providing us Mobile Dialer based SIP client which we are unable to configure as SIP gateway. We want to configure that Mobile SIP Di...
...call at different rates based on the destination and SIP trunk used. Billing is required per DDI and per client with the end result being a Xero invoice to be generated with the call breakdown per client/DDI. Currently have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant
I need a working FreePBX installation with three extension set (cisco 7975G) and one single sip trunk. I will give you SSH and HTTP access to the Virtual machine with a fresh FreePBX Distro installed. FreePBX have to be the latest with asterisk 13 13.14.0 and the Usecallmanager patch for the same. I will need the [url removed, login to view] for the
Consultancy on Skype for Business SIP trunking, and how to configure to get REFER in useable format to an upstream VOIP Server, the issue is the current S4B setup sends refer that results in null routing - example of issue is within the the below Refer and the Refer-To field as in - "REFER-TO:<sip:[url removed, login to view];transport=Tcp;maddr=10
Configure Asterix on an existing AWS instance. it's already installed but the configuration is not completed. Not sure ho...installed but the configuration is not completed. Not sure how long it takes to complete. I believe it's very short time if you know the scripts. The goal is to implement a SIP line that we got from Twilio. That's all for now.
...Microsoft C# WinForms platform. It'll perform VOIP calls on SIP Protocol and should be able to transcribe audio on the go (by stream audio upload to Google Cloud Speech API). Requested Features . Windows 7 (or greater) desktop application . Basic SIP operations . Register on a SIP Server . Make an outbound call . Answer a incoming call
Our company uses a hosted Asterisk server for comms. Staff a...sites. This new site, we have a Mikrotik router in place. I cannot get SIP to work through this router (one way audio) If I replace it with a modem/router, SIP works so issue is likely router configuration. Task is to log in to router, and fix configuration so that SIP calls work.
...something). The illustrations will range from simple animals, like 'clever cat' and 'mad rabbit' and single nouns ('mat' and 'map' are fairly difficult to do well), to verbs like 'sip' or 'nip', to short sentences like 'the cat went up the hill' or 'the dog bit the cat'. Before I define the project further or...
...documentation. We are looking to integrate A2Billing calling services on our customer service portal, where our customer support staff can manage customer accounts, create SIP accounts, and more. Full list of API calls will be attached to this project. Please note that we are looking for people who are experienced in A2Billing and looking for BOTH
I need some help installing and configuration from scratch goautodial system and also the configuration with the sip trunk provider for a telemarketing team, we need two types of campaigns, one for cold calls numbers from an uploaded database and the second one for sending IVR's from an uploaded database, calls and IVR's will be made to EEUU and El
...Phones should be using SIP Protocol. We will need to know how to use the TFTP Server to add and remove phones from the server. Currently we are using a TFTP Server which is provided by our supplier which is working great, but we have to tell our supplier the MAC address, Phone Extension, Extension Password and IP of the SIP Server which is not a problem
...like we will add pre-recorded voice that ""(we have vacancy for __ position, if you are interested then call us on this number)"" now that number should be diverted on our Mobile numbers and who ever will call back should connect with us. If are having experience of making such software & if you are Interested to make it for us then please do contact
...will be separated in two parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet