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    11,041 code agi web service asterisk server proiecte găsite, la prețul de USD
    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

    $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

    $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

    $1000 (Avg Bid)
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    1 oferte
    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

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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

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

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

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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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    ...some who have gotten this to work with other providers: [conectează-te pentru a consulta linkul URL] There are a variety of ways to interact with Twilio's service. The goal is to be able to tie FreePBX to it in a way that doesn't break upgrades to core, etc...Functionality should be externalized into

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    php mysql schedule creation 6 zile left
    VERIFICAT

    I have a server running with mysql and php, I am using it for asterisk based server for voice broadcast. I need help to resolve some issue and enhance its features. The out look will be like attached picture. Details will be discussed during chat.

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    Hi Aqs Y., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for some auto dialer based on asterisk

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    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

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    ...Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX server has a trunk to Skype Connect I need assistance

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    Hi Kingasterisk T., We have a problem with our asterisk configuration. as a brief when call directed to extension queue after ivr. there is silence and no result. caller cant reach to extension. as example we get this kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC

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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

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    I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign

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    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

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    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

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    8 oferte
    Need AI Experts 4 zile left
    VERIFICAT

    Hi All! I need AI expert who has a lot of experience in these fields. I want someone to make AGI algorithms for a revolutionary e-commerce project. Please write what you can do for AI parts of e-commerce project. Please bid only if you can prove yourself. Thanks!

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    AMD detection in Asterisk 3 zile left
    VERIFICAT

    I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.

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    im looking for someone to make landing page simple nice looking landing page have form and need image

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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

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    We have a call centre system using Asterisk. With Asterisk only supporting the following call distribution. We need to tweak the queue behaviour to reset agent idle time in two event, after agent has hung up an incoming call and after agent end pause. If you base on the ring stretagy of least recent, we can achieve the agent idle time reset on the incoming

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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

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    5 oferte
    Asterisk Problem 12 ore left

    hello i need to stream live calls from my asterisk more or so like chansky but i need to stream in demand and to wav files so i can play it in brower i am useing freepbx

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    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

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    We are looking for an Asterisk expert who has experience installing Kamailio or similar software to be used as an outbound proxy for NAT Traversal front ending multiple Asterisk/FreePBX servers. Some PBXs are having load spikes due to frequent device registration/authentication and NAT traversal functionality, so we're looking to install a solution

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    Add ADI input to Asterisk frontend. Thank you!

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    Need to develop end to end solution of call center.

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    freepbx FOP2 S-a încheiat left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.

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    freepbx fop2 S-a încheiat left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,

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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

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    Trabalho há mais de 25 anos em telecomunicações, estou procurando desenvolver um sistema de interfonia baseado em asterisk com acesso através de app (soft phone)

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    Current software versions: FreePBX: [conectează-te pentru a consulta linkul URL] Asterisk: 13.11.2 Project Description : . After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 2 fixed questions. We need to be able to define the Survey number and Name, then record the questions and select valid

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    Asterisk Voice IVR setup S-a încheiat left

    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

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    ...to establish an asterisk server and freepbx for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB

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    ...client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have to play audio file that inform user for call. When

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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.

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    ...for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inputs we trigger sip client to start call. (Gpio inputs must configurable in code) Sip client must have to play audio file that inform user for call.

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    Add the ability to enter sip trunk credentials to asterisk

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    ...than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server...

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    Build Re-seller Front-end for Asterisk Phone system to allow total control of asterisk via Html5 for management and configuring. To be about to add sites exts phone numbers etc

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    Build Asterisk Phone system for re seller with html 5 front end for management

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    Project for mula9211 S-a încheiat left

    ...now). We exist for over 10 years in NL and we have an asterisk based cloud PBX. We sell this as a service to companies and resellers. It is based using our own heavily customised linux distro. We also want to experiment with jingle (xmpp client) and SIP integration using a combi of sylk server and opensips. IMO a real challenge for experienced engineers

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    Hola No se si te recuerdas de mi hace mucho tiempo hice un trabajo contigo sobre asterisk, fue recibir pagos automatico usando el asterisk :) Si todabia estas trabajando como freelancer, estoy interesado en en tener ujn sistema parecido a [conectează-te pentru a consulta linkul URL] No se si lo conoces? lo estamos usando para call center, bueno cualquier cosa

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    asterisk security clean up S-a încheiat left

    fail2ban config and fix hacked phone.

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    Project for Viktor S. S-a încheiat left

    Hi Viktor S., I noticed your profile and would like to offer you my project. We need to install and configure asterisk/A2Billing for voip wholesale. Please let me know, when you can be available to discuss the details. Thanks, Nestor.

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    Asterisk developer S-a încheiat left

    helping in Project to finish product.

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    Server on Digital Ocean has been running great.. data spike at 3pm cst and then silence need assistance please.. could be log files full ? dont thing so login as: root root@pbx:~# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/[conectează-te pentru a consulta linkul URL] exist?) root@pbx:~# df -h Filesystem Size Use...

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    you have to install asterisk 1.8 on Puppy Linux make sure asterisk work with sip & iax both also wireguard & TeamViewer you can download OS from here [conectează-te pentru a consulta linkul URL] also if you want you can use Ubuntu or Centos you need to make iso and iso size must be under 500MB i can pay up to 100USD do not ask for more

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