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    1,313 freeswitch openbts zakázek nalezeno, ceny v EUR
    Freeswitch FusionPBX Support 6 dní left
    OVĚŘENÉ

    We have an codec error of FusionPBX Codec Passthrough issue mod_g729.c:102 This codec is only usable in passthrough mode! 2019-03-26 07:37:44.698244 [ERR] switch_core_io.c:1434 Codec G.729 encoder error

    €9 - €27
    €9 - €27
    0 nabídky

    CRBT integration using Freeswitch and CAMEL

    €9 (Avg Bid)
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    1 nabídky
    Freeswitch CRBT 3 dní left
    OVĚŘENÉ

    Help configuring Freeswitch for CRBT system. To make sure you understand requirements please start your bid with "CRBT" also please include link to similar projects.

    €491 (Avg Bid)
    €491 Průměr. nabídka
    1 nabídky

    i want to install call blockers in freeswitch astpp

    €134 (Avg Bid)
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    TRAINING ASTTP, FREESWITCH 3 dní left
    OVĚŘENÉ

    TRAINING ASTTP, FREESWITCH, create rates, carrier with any provider, trunk with any provider, like IDT, users, maintenance of server, security, etc

    €66 (Avg Bid)
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    I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on

    €24 / hr (Avg Bid)
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    If u have worked on the integration of both ASTPP and FusionPBX, then please contact us urgently as we have a job for u. They both run on freeswitch.

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    freeswtich Ukončeno left

    Freeswitch + ASTPP interface Full installation and configuration Trunk Setup Few Rates Configuration Test calls create user dialplan, prefix routing

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    FreeSwitch with ZRTP Server Setup

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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

    €1124 (Avg Bid)
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    I want someone expert in ASTERISK and freeswitch and ios to send calls over bluetooth but to use whatsapp or viber instead of gsm dialing to be smart enough to tell asterisk use whatsapp when dialing (send calls direct to whatsapp) android or iphone whatever they can fix

    €2498 (Avg Bid)
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    Hello, We are looking for someone to deploy a FusionPBX (Freeswitch) for us from scratch we even need everything to be migrated to the new PBX from our Asterisk.

    €182 (Avg Bid)
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    ...in right context, it is desirable, but not necessary, with experience and knowledge within • On premises deliveries of system and installation script • Telephony and SIP (FreeSwitch, Swyx (a German PABX) and eventually Asterisk) • Record keeping, eventually through a PHP framework • Access rights for different users, we work with an access token today

    €18 / hr (Avg Bid)
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    Voip expert with PBX (asterisk/freeswitch) and dialers (vicidial) experience to help deploying a special use case related to a voice bot product (not for human agents) A relationship with our low level developers. A long term job relationship is promised to whoever will fit the job

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    I need voip programmer to assist me with some projects in asterisk or freeswitch.

    €160 (Avg Bid)
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    15 nabídky

    ...your profile and would like to offer you my project: We have setup a fusionPBX/freeswitch server and are using sipjs as webrtc clients. Everything works fine except once scenario: extension A calls B, B picks up. So far everything is good. Now A hangs up, but freeswitch never signals the hangup to B, so B still seems to be on the phone while A is long

    €603 (Avg Bid)
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    ...accomplished on Freeswitch with mod_avmd. Perhaps it can be ported to Asterisk. [pro zobrazení URL se přihlaste] We need to have some dial plan variables for this module that allow us to adjust the tone duration and tolerances for amplitude and frequency. If one references the Freeswitch module

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    Looking for an expert to configuration FreeSwitch installation with ZRTP enabling. Need someone who is highly skilled around FreeSwitch, VoIP and Linux administration.

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    We use FreeSWITCH and opensips now. We want to redevelop a GUI for our Telecom System. If you are expert in this field, please happy to bid, We are looking for only experts now, Thanks.

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    Hello, We currently need an Erlang developer that also has knowledge regarding the Kazoo Voip Platform. Kamailio, freeswitch knowledge is a must.

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    There are bug module freeswitch Need fix

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    Hi I have a bts with openbts and I need a connection to one of the user of my bts to other bts transferring the information with an Gsm device

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    Need someone to tweak mod_amd [pro zobrazení URL se přihlaste]

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    We have setup a freeswitch / fusionPBX server and are using sipjs as webrtc clients. What we cant figure out is how to make presence work. We subscribe on the server, get an ok back, but we never get a notify with status info about the peers.

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    Project for Bilal A. Ukončeno left

    Hi Bilal A., I noticed your profile and would like to offer you my project: We have setup a freeswitch server and are using sipjs as webrtc clients. What we cant figure out is how to make presence work. We register on the server but we never get a notify with status info about the peers.

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    Client wishes to write a next generation layer on top of current free switch layer. The technology details are :Python, R, knowledge of free switch. (Very good coding skills) Experience level :2 to 3 years. Contract Duration : 6 months to 1 year. Start date : 1st March. Location : Pune Only locals please

    €1356 (Avg Bid)
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    Client wishes to write a next generation layer on top of current free switch layer. The technology details are :Python, R, knowledge of free switch. (Very good coding skills) Experience level :2 to 3 years. Contract Duration : 6 months to 1 year. Start date : 1st March. Location : Pune. Only local people please

    €1451 (Avg Bid)
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    ASTPP 3.6 running with freeswitch i want to listen live calls

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    Hi altr, I'm looking for someone to help me resolve an issue i'm facing on freeswitch. I'm using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure

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    ...faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar outra lingu...

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    Project for Arpit M. Ukončeno left

    Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload

    €354 (Avg Bid)
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    ...budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls

    €262 (Avg Bid)
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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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    We are looking full stack developers to build a telephony system based on engine Asterisk or Freeswitch. We need to build the whole system including a client area and manager interface. Experience on VoIP/Telephony proyect is a must. Please contact us for more details.

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    ...content must have the skill’s definition, advantage of the skill, where the skill is applied, why the skill is important in a business, etc LiveCode Attorney Flask iMovie FreeSwitch Google Cloud Storage Typescript Adobe Illustrator Scrapy Rust Qualtrics Survey Platform Snapchat Weebly Swing (Java) CSS3 Instagram API Brand Marketing Material Coating

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    Setup Kazoo - 2600hz - Cluster / Cross Data center Looking for an experienced company or freelancer to setup a Kazoo Cluster / Experience working with: FreeSWITCH Bigcouch Cluster Kamailio HAProxy

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    We need to Install OpenBTS remotely in Raspberry Pi 3 with bladeRF. - It is necessary that the OpenBTS work in Nuand bladeRF x40 with Raspberry Pi 3. - We need OpenBTS work in rogue mode to receive connections from every phone in the vicinity.

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    We need to Install OpenBTS remotely in Raspberry Pi 3 with bladeRF. - It is necessary that the OpenBTS work in Nuand bladeRF x40 (or other model) with Raspberry Pi 3. - We need OpenBTS work in rogue mode to receive connections from every phone in the vicinity. - We need OpenBTS to send 1 SMS message to every connecting phone and then immediately

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    Im setting up a network for VoIP. Am looking for an engineer that can assist in setting up the network with Opensip as the SBC and loadbalancing to 2or more Freeswitch servers. Security and firewall is a must in the network. Must have the ability to design the network. All equipment will be in the datacentre.

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    Freeswitch Admin Ukončeno left

    We are need for a Freeswitch Admin that can install & test the following services. Multi tenant, Presence/Chat, Real-time Users, Real-time Dial plan, call recording. We would like to hire somone to work by the hour to complete one project at a time. We will create the project, you will quote time, you will complete and document install process. We will

    €32 / hr (Avg Bid)
    Doporučené Neodkladné
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    Freeswitch Ukončeno 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@[pro zobrazení URL se přihlaste]

    €12 / hr (Avg Bid)
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    ...system but with other functions, probably more simple. Control on recurring payment, account, pricing, pre-created messages, list uploads, etc. 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

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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 ([pro zobrazení URL se přihlaste]) 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 c...

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    Hi. We use ASTPP ([pro zobrazení URL se přihlaste]) 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 c...

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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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    freeswitch Support Ukončeno left

    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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