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    18,220 sip authentication asterisk jobs found, pricing in AUD

    problem with authentication on my software..

    $220 (Avg Bid)
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    12 bids

    problem with authentication on my software.

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    ...the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling protocol) with the multimedia framework and NAT traversal functionality. This API is suitable for all types of systems including mobile handsets. Group video chatting

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    ...para integração com CRM via HTML Post, o cliente encaminha um link via html http://servidor/webservice/[login to view URL] e o webservice acessa o Asterisk via AMI user e password, faz a chamada, tenta encaminhar para o primeiro ramal disponivel, 3 segundos, não atendeu joga para o proximo até passar por todos, nesse meio tempo

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    I need a call forwarding solution set up in my aws account. I dont have any specific PBX software / service in mind - you can suggest the same. The intended use case is as follows: 1. Currently my service agent is taking calls from customers on their mobile phone directly. 2. Instead, I would like these calls to be routed through a server a) Let us say the service agent is on number 90000 ...

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    Looking for someone who is able to implement mac access authentication into a VERY SMALL python program, it needs to GENERATE these for the header.

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    i need to set up a system which will connect to a bulk list of mobile phone numbers in the usa and verify that they are active lines or disconnected + the cur...which will connect to a bulk list of mobile phone numbers in the usa and verify that they are active lines or disconnected + the current carrier. perhaps using voip's and asterisk ect. thanks.

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    need someone to finish home project PBX using freepbx 14. I completed the setup by the inbound calls fail. I have two test extensions I have one Sip trunk I have recordings for IVR i have one DID

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    ...recorded. I need to be able to query CDR to check all communication between parties + open recorded communications. Incoming calls will come from PSTN, SIP trunk, and all outgoing call through sip trunk. Script may be based on freeswitch as it’s an xml based solution, faster for administration....

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    Get 2 python scripts to work with my asterisk system

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    Our company Globetel Consultancy Service is based in Malaysia. We are looking for a candidate who can help to develop a SIP to viber/whatsapp gateway. The gateway should be able to convert the incoming voice calls over SIP and forward that through viber/whatsapp in order to complete the call to the receiver number. In addition to that it should be able

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    Imprescindible residir en España y poder emitir factura. Por favor enviar precio por hora. Aunque el precio seria por hora, antes de comenzar un desarrollo, el desarrollador, tras la presentación de cada trabajo a realizar, deberá de facilitar un volumen de horas cerrado. Pagamos 50% / 50% Se firmaría un contrato de confidencialidad y no competencia antes de iniciar el ...

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    I would like to develop a Xamarin with Linphone SDK Intercom system with this interfaces: VOIP Server: ASTERISK (Raspbx) 1) Intercom (Listed as "Portaria" in Mockup's). It will be a App installed in tablet placed in the front door of the building. User will touch the apartment number and a video calling will be performed. 2) Mobile (Listed as "App"

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    Hi everyone, I'm looking for someone who is familiar with C++ and with SIP to help me with updating this software and make some necessary changes. I need 3 changes made to it. 1) It's very outdated and needs to be updated with all the current packages and frameworks 2) Right now when an incoming calls comes in Outcall opens up a window in Outlook but

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    Details: -The App was built using the redux js library. -It integrates with Mind-body through the API. -Users need to be able to log in through the app and access data in Mindbody. -Users need to be able to book classes through the app. -The app is hosted on AWS Problem: Users are not able to log into the app since about 2 weeks ago. When users try to log into the App the connection fails.

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    VOIP Gateway 4 days left
    VERIFIED

    Hello, I need a reference design and prototype for a VoIP gateway Hardware with following ca...gateway Hardware with following capabilities 1) 1FXO-1GSM(SIMcard) port+1 LAN Interface. 2) OpenWRT firmware Based+Web GIU. 3) Be able to support L2TP/IPSec VPN. 4) Built in Asterisk Engine and Support. 5) Providing Design Documentation and tools. Good Luck.

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    This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset

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    i am working on an app that can provide unlimited calling from Canada to India( same functionality as Rebtel). I looking for someone who can provide pstn gat...functionality as Rebtel). I looking for someone who can provide pstn gateway solution in Canada with unlimited incoming. Any type of pstn gateway solution can be considered( ex. pri, sip......)

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    Looking for someone that has SIP/PJSip related experience to take our current PJSip plugin that works in NAT64 environments to work in a IPv4 only environment. The current code base is using this patch to work in NAT64/DNS64 environment: [login to view URL] This code works when on an IPv4 and IPv6 network, or IPv6 only network,

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    Imprescindible residir en España y poder emitir factura. Por favor enviar precio por hora. Aunque el precio seria por hora, antes de comenzar un desarrollo, el desarrollador, tras la presentación de cada trabajo a realizar, deberá de facilitar un volumen de horas cerrado. Pagamos 50% / 50% Se firmaría un contrato de confidencialidad y no competencia antes de iniciar el ...

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    Configure SIP server and post live calls gor each DID in database. Also we need to create and update inbound and outbound via API’s

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    I need to configure SIP server which supports API's for Inbound and Outbound also i need live call data based on DID

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    ...are building a freepbx server in Canada that can receive inward and outward calls and need some advising setting up a pstn gateway. For example pri line would be better or a sip trunk or if there is any other option ? Looking for a knowledgeable person who can get this project started. p.s. we will be posting some more jobs for the same so will need

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    I need a sample java program with asterisk integration that can connect to a freepbx server and can change basic configurations like adding or removing inbound or changing the outbound routes priority.I do have a basic setup of freepbx ready that i can provide. Quality work will be appreciated and can be re-hired for future expansion of the same program

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    ...VMs cease to work, or at best they work extremely slowly. On our Proxmox cluster, we are hosting 1 VM each for our website, syslog server, IMAP, SMTP, MySQL, MX1 and MX2, Asterisk, and our billing database. The IMAP, SMTP, MX and VoIP servers rely on the the MySQL server, and as far as I can tell, this reliance on the MySQL server will cause all of

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    Out store, [login to view URL] runs off of Magento community edition. A bug came up recently, which is preventing some customers from logging in, or changing their passwords. It is also preventing customers from checking out as a guest. This thread on [login to view URL] seems to outline the issue, and offers a potential fix: [login to view URL] However, the fix here either did not resolve our ...

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    Check urgent integration with Asterisk PAMI

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    Imprescindible residir en España y poder emitir factura. Por favor enviar precio por hora. Aunque el precio seria por hora, antes de comenzar un desarrollo, el desarrollador, tras la presentación de cada trabajo a realizar, deberá de facilitar un volumen de horas cerrado. Pagamos 50% / 50% Se firmaría un contrato de confidencialidad y no competencia antes de iniciar el ...

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    Imprescindible residir en España y poder emitir factura. Por favor enviar precio por hora. Aunque el precio seria por hora, antes de comenzar un desarrollo, el desarrollador, tras la presentación de cada trabajo a realizar, deberá de facilitar un volumen de horas cerrado. Pagamos 50% / 50% Se firmaria un contrato de confidencialidad y no competencia antes de iniciar el desarro...

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    ...if need be. I started doing this with Asterisk but then found rapid SIS. The dongle works (can send and receive messages, with raspbx, [login to view URL]) I have a configured raspberry pi which can be accessed via my VPN. I can send and receive messages from the pi using the asterisk commands. I can give you the VIP log-in

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    ...of my webcams. I Need a DLL Class that I can use as reference in my C# Desktop Application with the possibility to : "Make or Receive a Video Conference Call using H323 or SIP" In this class I would set properties like: VideoDevice (Select a webcam) AudioDevice (Select a microphone/line in) Volume Bandwidth Video BitRate Bandwidth Audio BitRate

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    We have a Vanilla SIPWISE Server Community Edition. We are looking for someone skills that can setup our SIP Providers and do test.

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    need to install and configure Kamailio and register several Asterisk server for Load Balance

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    asterisk[9194]: segfault at 6dd20000001a ip 00006dd34530b94a sp 00006dd22954fa10 error 4 in [login to view URL][6dd345293000+18a000]

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    asterisk crash can u fix

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    dear i have asterisk crashing problem , can u help i m using mnor kolmisoft

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    ...good voicemal application? Asterisk? Not bad if built rite. Take a look at Comsys voicemail applicaton features here: [login to view URL] Unfortunately, their pricing prevents many start-up service providers from using it. Your job is to study Comsys features and use Asterisk to duplicate these features

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    Hi, We're a training company and we require ssh access to various devices through a web browser - sounds like a job for Apache Guacamole! The easiest way for us to allow SSO to Guacamole is through the SAML 2.0 access provided by CAS. If we could get CAS and Guacamole working on a single Ubuntu box, that would be great! I've been working on this myself internally, but I need some help...

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    I run Ubuntu and have asterisk installed I need help with the following via ssh or however ... Finish configuring [login to view URL] (including allow= , deny=, alwaysauthreject=yes) Install any well known soft phone (such as Voiper) Change any default admin usernames and passwords in asterisk if any Exist Separate inbound and outbound routing Install

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    Hello, I recently added a TDM410 card into our pbx. I was able to add a MagicJack line to it by configuring a couple of dahdi cfg files. At this point I have a bad Echo. I want to to add another FXS line from an additional MagicJack dongle. I also have a Cisco IP phone that I need to configure it for one of the extensions of the PBX. So the help i need is: 1) Proper configuration of the Echo Can...

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    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

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    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

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    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

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    I want to install a fresh FreePBX/Asterisk on a DigitalOcean Server. Requirements: - Configure Firewall - Installing all the necessary admin, application, connectivity, dashboard, reports and settings modules. - Installing system sound languages for English, German, French, Spanish, Portuguese, Dutch, Italian. It is NOT required to configure any of

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    Hello! I have a newly started company in Sweden in which I need to utilize "BankID authentication". Full documentation to their service is available straight from their website, they also have a free testing service for easy testing of the newly devoloped software. The script I should be fairly basic, it shall only be able to call methods from

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    ...custom Outlook addin to connect to a SIP extension and when an incoming calls comes in it should give a pop up notification with the contacts info from outlook. It should be exactly like this: [login to view URL] but it should be able to connect to any asterisk SIP server. You can even use this addin

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    Hello I am looking for a developer to create a login and log out authentication system. After user logs in it takes them to a homepage. Need to also create a registration form if they are not sign in. If you need clarification, let's discuss more through chat.

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    Freepbx general support, particularly security, external sip phone connections like csipsimple not working, obi202 not registering, vpn connection setup, etc

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    I have successfully setup my angularjs laravel passport authentication. However I am having a problem to enforce it on my routes. Your task is to troubleshoot and fix the problem.

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