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    1,188 wikipbx freeswitch database jobs found, pricing in INR
    Freeswitch 6 days left

    I need to integrate a softphone with fusion PBX/freeswitch.

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    Hey Need to add a new feature ASTPP VOIP BILLING ASTPP use freeswitch pbx, but he generate diaplan. wee need add function pitch shift

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    Hello, We are based in Australia and looking out for someone who's good with FusionPBX as we are planning to migrate to it. We are ok with remote support but according to our times (Melbourn), please feel free to contact us at- info@[login to view URL]

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    ...Experience with Twilio, FreeSwitch, Integromat, etc. This will be a platform to offer other users to use the system to upload their contact lists or individual contacts and send ringless voicemails and SMS messages right away or schedule them for later delivery, they can be scheduled in bulk and/or for specific clients in the database Users need to pay

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    I need to connect 2 Ejointech gateways to my FusionPBX server. This is not a one time job. If I'm satisfied we can agree for a monthly maintenance

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    Hi. We use ASTPP ([login to view URL]) for billing. We need create a feature for transfer calls (Similar to FusionPBX *2). The ASTPP use xml_mod_curl for create the dialplan, once, yours need use the same engine to create the transfer calls. We already test the apps: bind_meta_app for capture the digits and works fine, but don't transfer to XML will make the transfer call (att_xfer). We ...

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    Hi. We use ASTPP ([login to view URL]) for billing. We need create a feature for transfer calls (Similar to FusionPBX *2). The ASTPP use xml_mod_curl for create the dialplan, once, yours need use the same engine to create the transfer calls. We already test the apps: bind_meta_app for capture the digits and works fine, but don't transfer to XML will make the transfer call (att_xfer). We ...

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    I need a coder that is very comfortable with Freeswitch to work with me on a project on an hourly basis. You will need to be able to create a conference room, play back a RTSP stream, as well as bridge calls to play back MP3/MP4 files. You will have to control the stream (rewind, forward, slow, fast, pause) as well as being able to store in the DB

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    We are looking for support to provide us with the following; Asttp Billing Freeswitch FusionPBX A2billing

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    We have Asttp Billing with Freeswitch we installed, it seems we are having an issue with the following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia

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    Looking for a good consultant with experience in Fusionpbx / Freeswitch who has worked in supporting production environment and has done customization.

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    We need to look for someone to manage our a2billing and freeswitch. Please install a2billing so that we can test and see your work.

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    Freeswitch Training that should cover freewitch basic, installation, configuration.

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    i need how to get 503 instead of any other failure code change it all to 503

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    Project Description Freeswitch is up and running Linphone for iOS and Android is up and running (able to receive a push, background calls) We have two issues: - When two clients practices call between together, the time of opening connection is very long. - The video quality is very bad if the clients move their devices. Please do not bid if you

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    ...install all the required software including freeswitch and kamailio customization needed 1.) using kamailio as proxy and freeswitch as the soft switch 2.) we already have a database and all the users must be authenticated with current database we have. database is a my sql server. 3.) the current mysql database stores client balances and it must be u...

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    Need a back end programmer for mobile VoIP app, should have a relevant knowledge, have done similar app is a plus, knowledge in FreeSwitch, Twilio, Nexmo and Other Voip related APIs

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    Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [login to view URL] which is a popular ringless voicemail platform in the US. However, we use other provider's APIs to do the delivery. This isn't cost effective for us. We need to operate our own delivery infrastructure. Is this something

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    We are looking for a technical person who has good telecom experience and can help us develop...[login to view URL] Http://[login to view URL] You can read about it and tell me if you can do it. We would ideally prefer some open source technologies like FreeSwitch trixbox etc.. Please bid only if you know about and able to help.

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    ...Management Tools ([login to view URL] with GLPI) e. Visitor Management Tools ([login to view URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools (Jabber) Basically the idea is to help new IT Managers and IT administrators to manage IT infrastructure using opensource tools to reduce the cost of

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    Looking for a good consultant with experience in Fusionpbx / Freeswitch who has worked in supporting production environment and has done customization.

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    Hi Nguyen Q., I noticed your profile and would like to offer you my project. We can discuss any details over chat. freeswitch

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    We need to build script via php or some other language where We can compare the hang up call + live Call acd and based from configuration we can hang up the call

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    ...changes to this softswitch and wants to integrate both so that calls from FusionPBX can go through ASTPP billing and ASTPP will bill the calls. You must be knowledgeable in Freeswitch as the system is a combination of two popular programs merged together. Start allowing registration of devices, providers, authentications etc – this means one can configure

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    I need a little help please. Restarted all my servers and now sofia gateway status doesn't show me any server. I have ASTPP installed with a cluster of Freeswitch servers. Error message I get is: astpp [ERR] switch_odbc.c:368 STATE: HY000 CODE 2003 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver]Can't connect to MySQL server on Please contact me

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    Hi, I have an installation of FS/FusionPBX on 64x Debian 8.10 on public IP, hosted by DigitalOcean. It used to work to an extent: phones register, calls were fine, ZRTP however never worked reliably (problems with key exchange, possibly wrong networking setup on the server) tasks: -teach me basics of admining the server (assume I know nothing, because I don't), like restarting processes, serv...

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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 to view URL] Eventually another

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    Setup Twilio voice bridge for bigbluebutton production server. Number taken, free switch configured, freeswitch/public/[login to view 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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    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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    account as a routing gateway I want to make same 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

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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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    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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    ...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 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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