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    14,616 asterisk pbx proiecte găsite, la prețul de USD

    Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!

    $200 (Avg Bid)
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    2 oferte

    Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,

    $250 (Avg Bid)
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    3 oferte
    Project for Anton F. S-a încheiat left

    Asterisk script

    $135 (Avg Bid)
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    1 oferte
    Server FreePBX Asterisk S-a încheiat left

    Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.

    $188 (Avg Bid)
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    Project for Mucahit T. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+407...

    $1000 (Avg Bid)
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    Project for Macovei M. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+407...

    $250 (Avg Bid)
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    1 oferte
    Project for Ionut R. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+407...

    $1000 (Avg Bid)
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    Project for slk S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de la RDS

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    Project for MikeRRR S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la RDS

    $30 (Avg Bid)
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    1 oferte
    Project for ionut0709 S-a încheiat left

    Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.

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    Specialist/Suport IT S-a încheiat left

    2-5 ani experienta in suport nivel 1 si 2 pentru sistemul de operare si aplicatiile Microsoft Cunostinte LAN, VPN, WLAN si Remote Connectivity, TCP/IP, la nivel de instalare si troubleshooting Cunostinte de diagnoza hardware si reparatii minore pentru laptop, desktop, imprimante, monitoare. proiectoare, solutii de videoconferitna, instrumente backup, alte periferice Cunostinte si experienta su...

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    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 - $3000
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    FreePBX Project S-a încheiat left

    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

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    Asterisk PBX -IVR 6 zile left
    VERIFICAT

    configure asterisk on ubuntu SIP Configuration ( incoming outgoing) IVR Menu (welcome menu, directory menu to (call groups) Create Call groups Call back request in case agents are busy for over 5 minutes) Enable Call transfer call record call waiting call transfer call hold tunes

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    Prezentat
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    asterisk pbx 5 zile left

    configure asterisk on ubuntu SIP Configuration ( incoming outgoing) IVR Menu (welcome menu, directory menu to (call groups) Create Call groups Call back request in case agents are busy for over 5 minutes) Enable Call transfer call record call waiting call transfer call hold tunes

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    Prezentat
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    i have a vici server and i am trying to install the webphone but i need to upgrade from asterisk 11 to 13

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    Asterisk Campaign Dialer 5 zile left
    VERIFICAT

    We need software that has a web interface, we can send a voice campaign and make a survey, and get the results.

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    Hi there, about a week ago you helped me with an Asterisk 13 PJSIP project. It is working fine, but one of my phones is no longer authenticating and may have some incorrect codec setup. Would you be available to remotely diagnose and fix it? If so, do you want to do this on your hourly rate, or fixed price? Myles

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    Development of a SIP to Whatsapp gateway required. Either standalone system or module for Asterisk/Freeswitch.

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    PBX Exchange/VOIP Application 4 zile left
    VERIFICAT

    I need my own PBX/Voip system like Vonage , Twillio dot com or Magicjack. This is for the call center. Need developer who can develop that. Must have prior experience doing this. Must have similar portfolio. The main requirement of the system:- 1) Outgoing/incoming call 2) Record the calls, track stats like duration of the call, response etc. 3) Software to access the stats, allow, block num...

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    Asterisk PBX billing system 3 zile left
    VERIFICAT

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. Min. experience required: 1) Asterisk PBX - CEL 2) AstPP billing software 3) LibreOffice Calc 4) MySQL DB 5) KVM qcow2 N.B.: What is experience ? -- It is when You have already worked with it... You don't have to google around ! For invoices and Credi...

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    hello, i have 2 ready freepbx servers + php code to fire calls throw asterisk api i need to add trunk on 2 freepbx + run my php code on it

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    i need help with my AGI dialplan ,if you complete project will give you more work for same, my budget is 50$ for this project .

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    I have a asterisk phone system setup working using Magnus Billing. I am seeking a reliable person that can update it and make changes as I need them.

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    Invoicing with Asterisk 1 zi left
    VERIFICAT

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply the background images for the LibreOffice Calc Spreadsheets. We use a small spreadsheet with background image for the "center part" of an invoice-template and an interchangeable "top" and "bottom" part on the sam...

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    rtp change 1 zi left
    VERIFICAT

    s it possible to forced change RTP IP and Signalling IP by using asterisk and / or other application ? example ; 1. Source RTP and Signalling IP is [conectează-te pentru a consulta linkul URL] and trigger a call to asterisk 2. Asterisk forced change to [conectează-te pentru a consulta linkul URL] and send the calls for termination. please let me kow Thanks

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    Hi Henry T., I noticed your proposal for project https://www.freelancer.com/projects/Linux/SIP-Whatsapp-gateway/details Can you please do the same work for me? I need VoIP Whatsapp gateway module for Asterisk or Freeswitch.

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    Asterisk server docs We need below points on Asterisk server. 1. Call will be generating from our web app by a user from his browser so we need the correct codecs to be setup on our asterisk server. 2. Once lead picks up the call then we check if it is a human or an answering machine (AMD system) If it is a human then the call will be connected to browser and if it is a machine then we simply hu...

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    details will be sent in a separate file. this is a custom listing.

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    idempiere 1 VCPUS RAM 3 a 4 Disco 20 ssd Linux Ubuntu 18.4. Segunda instância seria um banco de dados MySQL Duas instancias para site para site configurações mais básicas possíveis Uma para e-comerce 1VCPUS RAM de 3 a 4 Disco de 20 ssd Linux E uma para sistema asterisk Lembrando que todas as instâncias tem que ter o sistema de hibernaç...

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    Invoice templates S-a încheiat left

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply interchangeable background images for the LibreOffice Calc Spreadsheets. We need 3 small spreadsheets with background images of "top - center and bottom part" of an invoice-template on the same page but the client only sees the whol...

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    1 oferte
    Billing software -- 3 S-a încheiat left

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply interchangeable background images for the LibreOffice Calc Spreadsheets. We put 3 small spreadsheets with background images of top - center and bottom part of an invoice, on the same page but the client only sees the whole invoice, he doesn&#...

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    I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX [conectează-te pentru a consulta linkul URL] that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: [conectează-te pentru a consulta linkul URL] [conectează-te pentru a consulta linkul URL] [conectează-te pentru a consulta linkul...

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    I want to build an optimized image of the latest version of Asterisk for the latest Alpine Linux stable. The final product is the Asterisk compiled and tested with load in an AWS server and Docker. The system should have the codecs g729, opus and SILK compiled. The AWS image should be able to run without docker The Sources should be available to recompile minor updates

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    We have bought [conectează-te pentru a consulta linkul URL] sip trunk with FPBX. Now we need to deploy the pbx on a cloud. I understand the configuration but need someone with experience to guide us. The main work is to configure outgoing routes, maybe incoming routes. Once that is done we want the application to communicate with the server for phone calls (this we will do ourselves). I need a d...

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    voip - setup dial pattern S-a încheiat left

    i need some one to help me set up dial pattern for my voip , i have to outbound routes one for international, and one for local just need someone who understand how to set it up correclty I am using vitalpbx (asterisk)

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    Odoo developer -- 2 S-a încheiat left

    I need Odoo developer familiar with Asterisk distribution like Issabel , freepbx ...etc .

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    Invoicing software S-a încheiat left

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply interchangeable background images for the LibreOffice Calc Spreadsheets. We put 3 small spreadsheets on the same page but the client only sees the background image ! Before being uploaded to the server, the spreadsheets with background images...

    $402 (Avg Bid)
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    12 oferte
    3cx PBX set up S-a încheiat left

    Hey I have a 17 user environment that I need help setting up. My preference is 3cx and I will be hosting it in google cloud. We have 17 users spread over 2 sites currently, however, we have 32 sip numbers. Each employee has a direct line, then there are 2 receptionist phones (as well a 1 wireless phone for when the receptionist has to leave her desk) and the rest are on standby for expansio...

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    Hi all, I am looking for a system admin to install A2Billing on my CentOS 7 server. I have tried installing it using the following guide but it did not work: [conectează-te pentru a consulta linkul URL] I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,...

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    We would like to connect one Microsoft Teams user to a DID on an Asterisk box via SIP

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    website development S-a încheiat left

    Still looking for a salution and willing to pay to get this resolved permanently: Hello, I have a CLOUD BASED PBX it is NOT LOCAL and I have it linked to my pbx I uploaded the config file using TFTP and the Phone Does Register ( BUT ONLY WITH A LOCAL IP ADDRESS ) I am able to make out going calls to any number and to Extensions within my PBX but I am NOT able to receive calls when I call the exte...

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    Freepbx installation S-a încheiat left

    Freepbx 15 installation on DEBIAN or CENTOS OS. I tried many times but it never worked (calls drop after 32 seconds, or I hear nothing) I even would like to use API REST or other module to call url when incoming call to specific number, then this url respond with number that asterisk needs to callback.

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    Asterisk AMD implentation S-a încheiat left

    hello, we are looking to implement AMD on asterisk using give below case 1- We will send call from freeswitch to asterisk 2- Asterisk will send call to route. 3- once call is answered by called party 4- asterisk will run AMD 5- if AMD is detected the asterisk will hangup 6- if live person is detected then continue the call Thanks

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    Invoices, Bills, Templates S-a încheiat left

    To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply background images for the LibreOffice Calc Spreadsheets. We put 3 spreadsheets on the same page ! Before being uploaded to the server, the spreadsheets with background images, including all calculations , must be changed to pdf with open sour...

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    Project for Leandro D. S-a încheiat left

    Ciao Leandro, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

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    Project for Daniele M. S-a încheiat left

    Ciao Daniele, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

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    Hello, I want to build API [Java or php or C/C++] to call to Asterisk Server by using Android also from Windows [Browser]. The API will be one for both Android as well as Windows browser. I have purchased Asterisk SIP server users [Two number] and they gave me two extensions alongwith Server IP and password. Will shore the details after award the project What I need a API to integrate with Androi...

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    I have a set of c# asterisk applications compiled with Visual Studio that communicate using TLS 1.0 that must be recompiled to be deployed using TLS 1.2 and TLS 1.3. I am looking for and experienced Asterisk and C# programer to compile and assist in testing the deployment.

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    Having servers preinstalled with both Freepbx and Vtiger on Windows The needed is below : Integrating FreePBX based on Asterisk and Vtiger 7.x to help achieve the below: Click to Call from CRM • Incoming Call Popup with Contact/Lead/Accounts Details • Call pop up shows Previous Description • Call logs with all details • Create Account , Lead ,contact, Task...

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