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    1,373 freeswitch kamailio tugasan ditemui, harga dalam USD

    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    Hi, i would like to cooperate with an expert on FreeSwitch configuration. Just to make it straight, I don't plan to use any web interface as a plugin extension like FusionPBX etc. I want configurations from the filesystem. Currently, I just have two questions. But the person that will help me on those items i plan to hire with a monthly consultation

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    ...from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, 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

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    ...same VOS3000 softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work you can use any good Engine like asterisk,FreeSWITCH,Kamailio,OpenSIPS,etc More info check Attach file here is all screen shoot i need a complete wholesale Softswitch with a web management panel If you can make it please bid i

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    ...each user sign up and billing system ready for online payment 3) Integration with CRM. Technology expertise 1) The engineer must possess good knowledge of Asterisk and Kamailio system 2) The engineer must have an in-depth knowledge of SIP layers 3) The engineer must be able to understand and troubleshoot systems such as FreePBX and A2billing instantly

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    ...Now I'd like to develop a Voice Application using freewsitch for Voip Telephony. First I wan't to discuss what is the best architecture (Hardware, SO, Freeswich Version, freeswitch modules, failover, high avaliability), Install this Environment in my CloudServers, and then create a custom lua script that connects to my Artificial Inteligence API.

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    I'd like to create a voice application using google voice api with freeswitch.

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    I am looking for a person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE)

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    Hi i am looking for help on Big Blue Button Server setup, customization and branding along with performance optim...Blue Button Server setup, customization and branding along with performance optimization Knowledge of following Technologies may be required: Ubuntu, Web RTC, nginx, red5, FreeSWITCH, tomcat7, redis, Meteor, turn-stun coturn turn server

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    I need a fax sending mechanism using FreeSwitch to be installed and configured on ubuntu server. It should be accessible from my test node.js project.

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    I need you to develop some software for me...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. Please include the past project info.

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    ...telecommunications and IT conusltancy business. Our business name is Asterswitch, coming from the two main platforms we offer support/consultancy with (i.e. ASTERisk and freeSWITCH) and it is a one man team. Our business provides help with PBX and network solutions. The main objective of the logo: - must be a simple, elegent, professional stamp - must

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    ...any tool for load testing of free switch for audio conference 11) Deployment of multiple instances of freeswitch using load balancer Optional skills 1) Free switch video conference via Web RTC 2) Kurento media server 3) Integration of freeswitch with homer for live dashboard, RTP Packet analysis...

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    I'm looking for programming help with [login to view URL] and Kamailio. Must have experience with SIP and preferably experience with [login to view URL] library. Will need 1 - 4 hours per week for 3 - 5 weeks.

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    KAMAILIO SIP + MEDIA PROXY WITH SIP CAPTURE INTEGRATION INSTALL INSTRUCTIONS + CONFIG FILES + TECHNICAL SUPPORT We need a KAMAILIO CONFIGURATION accepting calls from one NIC with public IP address and redirect passing the call to 3 different SIP providers connect to 3 different NIC’s with “local” IP’s (load balance - round robin). Media must be redirected

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    We need a standard single zone seven server cluster installation of Kazoo. Full setup of Kazoo with working modules

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    ... Port test (tunnel): rtmp://[login to view URL] red5: [login to view URL] useWebrtcIfAvailable: true /opt/freeswitch/etc/freeswitch/sip_profiles/[login to view URL] (FreeSWITCH) websocket: X.Y.Z.72:7443 WebRTC enabled: true /etc/nginx/sites-available/bigbluebutton (nginx)

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    I am looking for a freelancer to help me with my project. The skills required are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers

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    ...Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over. Youll need: NAT & Voice skills on Cisco So far, I've been successful in getting

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    ...system. We 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 Twilio, or FreeSwitch etc..If you Already have completed a Ringless voicemail drop system please bid. Please include the past project info. Thanks Skills required: Asterisk PBX, Telecommunications

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    ...caller calls in. Our PBX platform is built on top of FreeSWITCH. We have a developer already that is very knowledgeable with FreeSWITCH and is going to help with this project. Here is a page that describes what we are trying to do. [login to view URL] Freeswitch does have an Event Socket Library that will allow integration

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    I need someone with experience configuring freeswitch to help me configuring an inbound DID

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    WE NEED AN ASTERISK EXPERT TO INTEGRATE HYLFAX AS AN OUTGOING ONLY BROADCAST FAX SERVER. Fax Broadcasting is sending an identical fax to multiple recipients 1. SOF...MUST INCLUDE A NUMBER RECEIVERS CAN CALL AND DELETE THEIR NAME FROM OUR FAX LIST BY ENTERING THEIR FAX NUMBERS. MUST HAVE EXPERTISE IN: C, C++, AWK, VOIP, ASTERISK, PYTHON, FREESWITCH.

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    Skilled Technichan and developer FreeSwitch.

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    I'm looking for a tech who already has completed a Ringless Voicemail drop system. We ar...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

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    ASTPP installation and configuration need such as below task: 1. Install ASTPP 2. Security: a. Apache Authentication b. Secure Freeswitch c. Secure Portal d. Fail2ban. 3. Install letsencrypt Optional, If you have more experience we need some extra customization for call recording features on customer CDR page which can listen and download it self

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    Hello we have a LTE EPC with titan HSS, its working only for the data now and we want to intergrate a opensource IMS like Kamailio IMS server for the Voice Over LTE.

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    I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail fea...about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch

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    ~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at setup cl...Hong Kong - 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).

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    ...report to the existing SMS system as there is no report there. Test the system to make sure all components work well before we can release it. You must be knowledgeable in Freeswitch as the system is a combination of two popular programs merged together. We only want developers who are knowledgeable in FusionPBX and ASTBB and have worked on these systems

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    ...outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: 1. Perform NAT traversal on any incoming or outgoing SIP connections using the "nathelper" module 2. Simply relay the SIP connection to an EXISTING PBX server (3CX). ***Kamailio should relay ALL SIP requests

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    Looking for FreeSwitch/Kamailio Installation 3-5 years of experience can install and setup Freeswitch/Kamailio from scratch, Support to resolve initial issues/bugs . VOIP phone features HD voice Code G7.29 support HD music on hold Call forwarding Three way conference Blink transfer Hunt group Follow me (forward to another number) Auto Attendent (Answer

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    i need to install and cnfigure freeswitch with GSMOPEN module, to allow make calls while connected to internet. and have full control over how calls will be handled( max duration perday, delay between calls, maximum unique numbers per day or month ...) . and a a web interface to manage sim cards (topup, check credit...). also a script to donwload content

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    Looking for FreeSwitch/Kamailio Installation 3-5 years of experience can install and setup Freeswitch/Kamailio from scratch, Support to resolve initial issues/bugs . VOIP phone features HD voice Code G7.29 support HD music on hold Call forwarding Three way conference Blink transfer Hunt group Follow me (forward to another number) Auto Attendent (Answer

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    I have Bigbluebutton 2.0 on a VPS. I want to migrate it to Amazon EC2 or Google compute ...engine. Help me purchase through either of them that is affordable and migrate current BBB 2.0 from existing VPS to new EC2 or compute engine completely. And help configure freeSwitch for inbound calls via IDD trunk and voice prompts. Send me fixed price quote.

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    Looking for FreeSwitch/Kamailio Installation 3-5 years of experience can install and setup Freeswitch/Kamailio from scratch, Support to resolve initial issues/bugs . VOIP phone features HD voice Code G7.29 support HD music on hold Call forwarding Three way conference Blink transfer Hunt group Follow me (forward to another number) Auto Attendent (Answer

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    6 bida

    Looking for FreeSwitch/Kamailio Installation 3-5 years of experience can install and setup Freeswitch/Kamailio from scratch, Support to resolve initial issues/bugs . VOIP phone features HD voice Code G7.29 support HD music on hold Call forwarding Three way conference Blink transfer Hunt group Follow me (forward to another number) Auto Attendent (Answer

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    voip app build from scratch for demo see this app [login to view URL] i ...app build from scratch for demo see this app [login to view URL] i need a clone like this app kamilio and freeswitch server experience preferred

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    Can you make Linphone wake up with flexisip or opensips or freeswitch push notification and receive incoming calls or sms ? Freeswitch and linphone is already up and running my budget is $500

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    ...communication between parties + open recorded communications. Incoming calls will come from PSTN, SIP trunk, and all outgoing call through sip trunk. Script may be based on freeswitch as it’s an xml based solution, faster for administration....

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    i want to integrate flexisip as front-end with freeswitch as back-end. however i do not know how to configure the flexisip to communicate to the freeswitch.

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    Hi. Looking for somebody to install Freeswitch with a simple web based interface to admin it. Nothing complex is needed - user and call groups creation and deletion. ZRTP is a must. Please contact me if you have any questions.

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    need to install and configure Kamailio and register several Asterisk server for Load Balance

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    Integrate Freeswitch or asterisk with Pipedrive. The freeswitch is my first option, but we can work on asterisk. On asterisk I already has RestAPI. On freeswitch we need buildtin. I want my software show on plugins of pipedrive. The integration want call from pipedrive, record the call details on pipedrive, etc.

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    We need help to integrate flexisip with linphone and freeswitch. Flexisip is partially working with push and outbound calls, we just can't figure out why flexisip database is not storing sip user/pass. Also need help with redis server integration with flexisip

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    We need help to integrate flexisip with linphone and freeswitch. Flexisip is partially working with push and outbound calls, we just can't figure out why flexisip database is not storing sip user/pass. Also need help with redis server integration with flexisip

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    Freeswitch is up and running (able to convert SMS to SIP messages and forward to Flexisip) Linphone for iOS and Android is up and running (able to to receive push, background calls and SMS) Our developers can provide some support for Mobile Applications and Freeswitch, they have limited knowledge related to push notification. - Flexisip must be installed

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    We are customizing fusion pbx with a billing system together. This has been done successfully although untested. Now few things remaining on the project: 1. We want to add API for international mobile top up or mobile recharge - this will involve adding security and other things that can stop frauds from happening. We will present a guide to the successful developer. We also need reports for a...

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    inphone integration with Freeswitch via Pushkit COMPLETED - App Pushkit integration - Server Push Integration RESULTS: - Currently we are receiving random results, some times app is able to receive calls when terminated or in background REQUIREMENTS: - App should be able to receive Calls & SIP Messages in background mode - Require help to troubleshoot

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    I see you have experience in VOIP. I need to install Freeswitch on a Centos server for use in my voip termination business. Please let's chat to see how we can work together. Don't know if you have any experience with device configuration. I will like to discuss that with you.

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