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    1,161 freeswitch interface pekerjaan ditemukan, seharga USD

    Chat later...jkafdhjaf;jga;jgja;jg;jafga;ssakjskf;gjas;gdks'g;gjjj;;jskjjkjgjasjjgaklljhdlaa;lIGldhgdKGHLDGHljfld;jLKDDGJDF']GJLKF

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [login untuk melihat URL] Eventually another

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [login untuk melihat URL] Eventually another

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    Setup Twilio voice bridge for bigbluebutton production server. Number taken, free switch configured, freeswitch/public/[login untuk melihat URL] setup. Need to add var/www/ TwiML configuration to enable the audio bridge. Respons only if you know Twilio and BigBlueButton integration and can do the job within one hour online using any desk now.

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    Hi All Im using FreeSwitch today on my LAN running on windows. It works fine. However its flakey I have a Cisco 3945 with Voice/UCS license that I'd like to run a SIP server on IOS to replace this. I only use this for LAN, there is no PSTN calling. I need my LAN devices to register to the SIP server and also my 4G Mobile device (using Zoipher client)

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    Want to chat re: freeswitch

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    Dialplan Freeswitch Berakhir left

    Transmitir streaming via rtpm por freeswitch cuando la llamada es contestada.

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    Implement G.722.2 AMR-WB and EVS codec as FreeSWITCH module. Not a passthrough module, must actually be capable of transcoding to L16 (or whatever is considered the most native for 16khz stream in FS). Purpose is obviously to support wideband audio to wireless phones when supported. It will be our responsibility to license with VoiceAge.

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    Need ANSI C Programmer with FreeSWITCH module programming experience. Also, experience with AF_UNIX named sockets. Module needs to buffer sound bytes of audio from A leg into memory, package and deliver to another process upon silence threshold being met, and continue until timeout or message received from socket. You could start with mod_dptools

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    I'm stating voip system and calling card with soft-phone service. I want some one who can install freeswitch with fusionpbx billing solution.

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    Hello. We are Looking for Call Center Solution based on Freeswitch (Multitenant) (Not Vicidial or GoAutodial) along with Source Code. If anyone have ready, contact us and provide us demo. We will purchase instantly. For more information you can chat with us. Thank you

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    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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    ...to 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

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    Project for Aqs Y. Berakhir left

    ...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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    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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    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 untuk melihat URL] red5: [login untuk melihat URL] useWebrtcIfAvailable: true /opt/freeswitch/etc/freeswitch/sip_profiles/[login untuk melihat URL] (FreeSWITCH) websocket: X.Y.Z.72:7443 WebRTC enabled: true /etc/nginx/sites-available/bigblu...

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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 untuk melihat 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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    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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    VOIP/SIP Project Berakhir left

    ~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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    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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    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 untuk melihat URL] i ...app build from scratch for demo see this app [login untuk melihat 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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    freeswitch project Berakhir left

    ...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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    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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