Build asterisk interfaceproiecte

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    10,673 build asterisk interface 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

    $30 (Avg Bid)
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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.

    $150 (Avg Bid)
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    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 - $3000
    $1500 - $3000
    0 oferte
    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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    ...experience. We have a current very old custom build solution based on a Asterisk PBX with ISDN and want to move it to the cloud and provide new features for our team. Its very important that you have indept knowledge arround Asterisk PBX and integration with SQL / External scripts from within Asterisk PBX We can accept that you just provide the SQL

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    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    35 oferte

    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    35 oferte

    We need a server specialist that can install tonight (with information from our developer) an Asterisk VoIP Server on a CentOS. Next to installation and proper working of software we need you also to increase the security settings of the server. Budget $100. Must be done in one day.

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    Fix small issue with Goautodial 6 zile left
    VERIFICAT

    Looking for someone who are expert in fixing the asterisk server and goautodial server to fix some issues. I am trying to connect that server to external public IP behind the firewall/router. Somehow there are some errors such as " handle_request_register: Registration from '"1300" <sip:1300@[conectează-te pentru a consulta linkul URL]>' failed for '[con...

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    I have a Asteris 11.14.1 PBX. I want to use Bitrix24 CRM and log all incoming and outgoing calls in the CRM. I need to connerct Asterisk with Bitrix24 for inbound and uotbound calls

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    I am setting webrtc to an Asterisk server,I need an engineer with experience in networks and Asterisk webrtc configuration, in order to fix error net::ERR_CERT_COMMON_NAME_INVALID. I have a Goddady certification. Asterisk show TLS/SSL certificate ok.

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    ...} If I request this web address, Asterisk will: 1-) Call {CALL} via {CALLERID}. 2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service. - Need support over 50 calls at same time. - I have a server at digital ocean. Asterisk is not installed. - I have a SIP account

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    ...if I try to solve the pages does not update correctly the Vicidial/Asterisk MySQL tables. I need a correctly full working installation, the result should be an installable and fully working ISO or Virtual Machine harddisk with the working installation. Requires: dkms-dahdi, asterisk, vicidial, php7, MySQL, MariaDB, CentOS, and other of the same kind

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    asterNET modify sip trunk 4 zile left
    VERIFICAT

    asterNET modify sip trunk need a vb.net or c# to modify sip trunk in asterisk from a windows base software

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    We need to integrate an asterisk pbx with SuiteCRM if possible with freepbx or raspbx Integration means: - On inbound call open CRM calling contact information - Clicktocall from CRM - Logg inbound and outbound calls in CRM

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    Hello All, I'm looking an experienced Microsoft system admin. The freel...windows server 2016 needs prior to be prepared before the installation of Exchange server 2016. UC must be configured as well since we will also be hosting VoiceMail for an Asterisk VoIP system Note: Apply on this job only if you already perform such as job. Regards, Léon

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    We need someone that could write an interface to make sure that Asterisk and a Hotel Management software, Opera (that supports FIAS Protocol) can communicate, especially the billing of the calls, wakeup call, caller id.

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    ...following project: Hi, I read your gig and would like to share with you what kind of project we have in my company in Bulgaria. I am a little bit confused about PBX, VoIP, Asterisk therefor I will describe what we need as functionality. Right now in the company there are 3 hard phones, one landline number 00359 (0) 52 630 540 and one fax number 00359

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    Create a Survey FreePBX/Asterisk module to measure customer satisfaction levels for our call center agents services. Current software versions: Elastix 4 Here is what is needed. 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 1 or more fixed questions. We

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    I have a personal asterisk server for my self and kids and I see lots of messages like this. --STATUS: HANGUP on h from from ,,XXXXXX") in new stack Looks like some one trying to make calls but I dont have outgoing plans just a few extensions. I have fail2ban installed but it doesnt seem to be catching them.

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    ...fields ... the only field I have doubts about is the one labeled as follows: The difference between your server time and the [conectează-te pentru a consulta linkul URL] server time (in seconds)*: The asterisk (*) at the end of the field name corresponds to the following: * Please ensure your system clock is set to proper time and time zone. AuthorizeNet gateway will validate the

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    Hi! I need help with Freepbx CRM and asterisk API's: I need customers to be able to make a call from the CRM on the browser with a click on a button using an API, not with Zulu software or plug in.

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    Asterisk Dialplan S-a încheiat left

    I have asterisk server with whitelist number and make any number hit my asterisk to pass on this whitelist and if number in whiltelist it pass if not it hangup . now i need to add some options for non whitelist numbers like playback press 1 to continue and if he press add it to my whiltelist also need to send 503 code when call hangup after not passing

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    we need Asterisk Call Center System developer to work with us project base .. with full experience of : - Basic Call System - Call Center System - Call Rating System - Call Details Reports - Branding The system for us .. - CRM Integration

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    Asterisk/FreePBX Genius! S-a încheiat left

    I need the full Amazon AMI Install with Freepbx tweaking in order to make this 100% functional. [conectează-te pentru a consulta linkul URL]

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    Trophy icon WebRTC help needed S-a încheiat left

    I use [conectează-te pentru a consulta linkul URL] v0.11.6 for my WebRTC project. Everything works well but there is no sound on reciever side. Only caller can hear. Asterisk 16 is installed on Centos 7. Need help to fix the issue.

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    Garantat
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    3 intrări

    We have 3 issues with this integration: 1- For the outgoing calls, the popup shows the trunk ID number instead of the dialed number. 2- When rece...When press on the call icon in the web browser, the popup show up but the extension is not calling. Project is on high priority, Please only bid if you have experience in Asterisk with Zoho Integration.

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    ...see: The numbers of fields that has to sign on the bottom (fixed) Only the fields that has to sign with a yellow background Red borders on the bottom and an asterisk if required The text will fit inside the cell if longer will be resized. If the user use the desktop, can sign using the mouse. When the user click the save button, the

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    Prezentat
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    I have an asterisk server: Asterisk 1.8.20.0 built by palosanto @ [conectează-te pentru a consulta linkul URL] I need a php based script which will popup a browser window for every incoming call & show information based on QUEUE & DNID. Budget: $50

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    want to set up my very own SIP server /VoIP system with a VPS server i will buy for the project....want to dial 800 type numbers thru MicroSIP, no web interface is needed. I need someone to fully set up the system and explain everything to me. Specifications will be sent when project is accepted.

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    need to edit the [conectează-te pentru a consulta linkul URL] from vb.net or c# urgent

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    Need someone to set us up with a Virtual PBX or cloud call center. We want to see calls in real-time and put callers in queue. Thank you!

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    1 oferte

    Need someone to set us up with a Virtual PBX or cloud call center. We want to see calls in real-time and put callers in queue. Thank you!

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    Asterisk Developer S-a încheiat left

    We are facing issue where Caller did not hear voice from agent. This issue happened random time and mostly occur in the morning (Malaysia). below the detail: Version: Asterisk 1.8 & Centos 5.9 IPPBX server [conectează-te pentru a consulta linkul URL] Digium Gateway [conectează-te pentru a consulta linkul URL] And we have wireshark capture. PABX sent 3 SDP traffic to the phone. Acco...

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    I have some trunks (inbound call numbers) I need to know which ones to call, call pop up should show the trunk name that the client called. Vtiger 6.5 or 7

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    Part time support work for existing Asterisk customers, customization of recordings.

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    ...project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have to be specified)

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    SETUP VITAL PBX Asterisk S-a încheiat left

    I have installed and configured a system with VITALPBX, the extensions work but they are not able to configure the external line. I'm looking for someone who helps me and teaches how to configure it

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    Hello all. We need to connect 2 databases between an asterisk and a softswitch. Thanks. Fredy.

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    Configure FreePBX 14 on Vultr VPS. Set up Endpoint manager, configure firewall and help with phone provisioning

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    SNMP and Syslog Install S-a încheiat left

    ...Professional Windows 10 Enterprise Windows Server 2012 R2 Windows Server 2016 Peplink Router Balance Series Pepwave SOHO series Pepwave Access Points Proxmox cluster Centos OS asterisk FreePBX Arista Switches Nortel/Avaya Switches FreeNAS Cannon MFC8080cw printers Xerox 6605n printers Tripplite UPS with SNMPWEB card installed APC Automatic Transfer Switch

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    Hi, I have fully installed Asterisk 16 and working webphone based on sipjs. Audio on that setup are working, what I need is a someone who reconfigure it and make for me working videochat on this. It could be sipjs or sipml5 i dont care. It have to works audio call _AND_ video call

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    ...WEBSOCKET to work with Asterisk. (ej, mizu, duobango, kurento) 3- Tell me clearly the additional costs in scripts or licenses. 4-Demonstrate and make an example of a WEBrtc client in order to see the functioning of the client web server the voice communication. I need to know how to implement voice communication between the asterisk server and web pages

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    natural language S-a încheiat left

    VAT natural language por asterisk

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    Ho la necessita di registrare un telefono cisco cp-6945 su un centralino grandstream ucm6204. il telefono ha già caricato il fw sip. Il telefono non permette la configurazione tramite web access ma solo tramite file XML

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