SIP integration
Hi, I have a Ruby (3.1.1) that needs to send calculated UDP payloads via SIP (RFC 3261), an example payload being: PL7QgaX1F1PI110195gsm:10 Sending via Ruby is...
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FIX Linphone SIP Audio Issues after upgrading from 4.4 to 5.0 (Cordova/Ionic)
FIX Linphone SIP Audio Issues after upgrading from 4.4 to 5.0 (Cordova/Ionic) Device issue is Lenovo M10 (some models do/dont work) THe far end can hear audio b...
Read moreOpenSIPs Admin
mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Di...
Read moreAvaya pms-link is up but guest name are not changed on cm
Avaya pms-link is up but guest name are not changed on cm
Read moreANDROID VIDEO CALL APP- BUG FIX
Hi guys, I have a video calling app that runs on LINPHONE CLIENT. play.google.com/store/apps/details?id=com.schonell&gl=SG The only function of the app is to re...
Read moreAsterisk [SIP] based Call Center Solution
Hi I am looking for a developer with extensive experience in Asterisk to build a call center based solution over SIP. Only experienced developers who have worke...
Read moresocial media handler
i need a person who can handle social media for me to promote my coaching classes and export business.
Read moreСоздать Predictive Diller на CFD для 3CX
Необходимо пользуясь всеми доступными системами 3CX создать правило обзвона клиентов, с забором номеров из файла и обзвоном по интелектуальной системе свободных...
Read moreEntwicklung Nextcloud APP
SIP Client als APP für Nextcloud mit JSSIP library Folgende Funktionen sollen unterstützt werden: Zum Wählen ein Tastenfeld 0-9 *,# Annehmen und Auflegen mit DT...
Read moreSetup Kamailio
We need help installing a Kamailio SIP redirect server. See specification in PDF Thanks /Dan Erik
Read moreBuild WEBRTC to SIP SDK for developers to Connect + Documentation
Build WEBRTC SDK for developers to Connect + Documentation need to build a WEBRTC SDK (gateway to SIP.js ) but not entirely sure how to build a SDK so users can...
Read moreAutomatic Call Distribution Software (ACD) -- 2
I am looking for an expert who have already worked on such project for forwarding and ACD Setup for call ceneter. Where i can route multiple calls to multiple c...
Read moreSIP - 2 way voice communication between 2 local Linux computers
Hello, I'm looking for an Asterisk / SIP expert to build a 2 way voice communication between 2 local Linux computers. Both computers run on the same network. We...
Read more2-way talk - voice communication in Linux command line
Hello, I'm looking for a Linux EXPERT with a solid coding skills to build a command line script for two ways voice communication. I need to run a script on Linu...
Read moreSip Proxy Server / Sip Forwarding
I need a solution to forward all calls from one sip domain to another sip domain without handling the media, the solution needs to be able to cope with up to 10...
Read moreCustomize Linphone
We are looking for someone who has experience with Linphone to help us build the code on our machine and customize Linphone. See the source code and doc here: h...
Read more3CX - Voice communications not clear, We suspect this is relating to PORTs not properly set
Using 3CX v18 (GCP HOSTED) Local SBC, 2-3 Yealink Phones already provisioned On-premise router, Fiber
Read moreLinphone and ZeroTier integration and customization
Dear all, I am seeking mobile APP and software developer to integrate ZeroTier into LinPhone softphones (Mobile APPs ANDROID & iOS & desktop applications for Wi...
Read moreOpenSIPs Training for Staff
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want t...
Read moreVOIP engineer Asterisk specialized
VOIP engineer Asterisk specialized The SIP provider capture files must be understood for (Incoming calls + outgoing calls + sip trunk registration) This files c...
Read moreKazoo VoIP CUSTOM developer For Kazoo-SDK
Need to look at building out some custom kazoo-SDK to work with nestJS
Read moreOpenSIPs Training -- 3
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want t...
Read moreSIP Protocol / Handshake Backend FlexiSip Server Debugging Required
We have a WhatsApp-like app that we developed. It is much more secure than other chat apps. We have added secure audio and video to our app based on the open-so...
Read moreFreepbx Issabel
Need installation support for Free pbx, Issabel and sip configuration
Read moreI'm looking for a German SIP number / Voip; no dialer, no call-center software
Hello everyone, for a reputable company based outside of Germany, we are looking for a German phone number with out & inbound function. SIP/VOIP which can be us...
Read moreLinphone background calling problem Android
Hi, I have an android app based on linphone sdk-4.4, everything works fine, and app is able to receive the calls. but when you restart the phone and phone comes...
Read moreOpenSIPs Training -- 2
I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want t...
Read moreOpenSIP / Kamailio with CPS control
Customer has a few dialers using asterisk and sending call to a single provider. Combined call can goes up to 100-200 calls per second, but provider only provid...
Read moreSimple Kamailio Configuration
We are looking for a simple Kamailio configuration to act as a SBC of sorts. We have a SIP based fax server that isn't able to fail-over to the next peer if the...
Read moreIntegrate Linphone Sign up with Astpp
Good day Looking for a person who can integrate linphone apps registration with ASTPP open source
Read moreCustomize WebRTC Solution
Hello Everyone To do this, you must know asterisk and FreePBX and SIP and WebRTC. If you don't know either one, don't bid. My Goal: I gonna integrate webrtc sip...
Read moreIT Technician
Avaya (settings), fortinet, vmware, cisco. zktecho. active directory. windwos server, gpos.
Read moreSIP Audio converter (convert any audio to sip protocol)
We want a voip professional/developper/engineer to create a code to be able to convert audio to sip protocol and from sip protocol to audio. The code can be upl...
Read moreDiscord Call Bot (using sip trunk)
I need a discord bot coded that is able to do calls via a sip trunk. Also the bot should have call recording and muting users in voice channel.
Read morePython PJSIP,DTMF, gTTS,AUTOMATED CALL
Hi everyone, I'm looking for someone to make for me a python script that interact with VOIP Using PJSIP. The script must call a number and use gTTS ( Google Tex...
Read moreVoip and sip trunk guidance (FreePBX )
Need someone to be able to set up a FreePBX on AWS and integrate it with Twilio number. The most important thing is to be able to guide me through the process a...
Read moreSippy soft switch
I am looking for an expert who can install Sippy soft switch on my server. It's related VOIP business kindly contact me urgently. Thank you
Read moreIntegrate GCRates with Freeswitch
Hello, We need to implement CGrates with our current SIP infrastructure. The infrastructure in mind is an SBC server connected to a cluster of 2 Freeswitch. We...
Read moreVOIP-Techniker mit 3CX-Advanced Zertifikat gesucht
Wir sind ein deutsches Cloud-PBX-Unternehmen, welches eine 3CX-Cloudlösung für seine Kunden anbietet. Wir suchen dauerhafte Unterstützung für Installation/ Einr...
Read moreElectron Developer with Web RTC knowledge Needed!
Hi, We are looking for a professional developer who is an expert in Electron & WebRTC to debug an issue while answering calls on our electron app when used on a...
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