Opensip ldap asterisk jobs

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substitute ldap server protocol with http protocol. Currently using node.js and mysql. fetch data from http protocol now and want to swap it with ldap protocol. somebody here who can do this? want help and hire soon if doable

Looking for a Centos Guru who knows how to set-up, resolve issues especially with Active Directory. We have a Centos 7 VM that following some upgrades no longer talks to Active Directory. Need someone to look at the config, set-up and...the config, set-up and resolve the issue(s) so that a specific AD Security Group can login and then sudo -s. Need to retrofit any changes we've made (limited as precaution). One clear issue is below net ads info ads_connect: No logon servers are currently available to service the logon request. ads_connect: No logon servers are currently available to service the logon request. Didn't find the ldap server! In addition once resolved, to then document how an AD user that can SSH / sudo auto mount a volume SMB (Windows File Directory). us...

...from lower to higher versions within a given solution. Experienced in solving for content mapping, content integrity, cutover mechanics, namespace issues, etc. • Good understanding about version control (SVN/TFS/GIT) • Ability to build new Jenkins Job for the new AEM instances is a plus • Knowledge on the Adobe Experience Manager administration activities such as User permissions/creations with LDAP synchronization, sling setting status, creation of audit reports, dispatcher setup, code and configuration management and others. • Setup and maintain administration activities such as workflows, publishing permissions, roles, templates and components • Experience with Apache Web Server configurations. • Experience with Linux server environments. ...

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

I have a system asterisk 3g modem, I need to install for me a database for IMEI of dongle purpose for this operation is when i put a dongle with new sim in my system dongle change automatically it imei. Pkease confirm it is possible to do it.

...Freepbx's Zulu Mobile client (iphone/android). Right now the issue is: On a new server, I can use Zulu's PC client to do outbound/inbound to external calls, but with the mobile client, I always see this error below when I do an outbound external call, and the call will hang up itself, the mobile client out close it by itself too. I've tried this on both iphone and android client. Can any asterisk/freepbx professional provide me a quick solution for a fee? Full log: -- SIP/6566811234-00000005 answered PJSIP/90101-00000005 -- Channel SIP/6566811234-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> -- Channel PJSIP/90101-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-...

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of 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 B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from U...

... I need to replicate there backend using only asterisk. That requires implementing asterisk to my AWS server. To be specific , I need a developer to add asterisk to my server and develop a secure rest api that will prank call two people for you using asterisk in my server. Please use the website i linked above to understand exactly what kind of prank calling I'm trying to implement. After the call is made through the server it should record the call up to 30 seconds , 1 min or 2 min only and play and audio file in the end before the call ends. This audio file will be provided by me. This audio file will be saved via my google firebase could storage. After implementing asterisk and making the functionality work. I need a secure rest api to call ...

We are looking for a passionate and talented VoIP Engineer to join a multicultural team (including French, Russian and Romanian) and help us tackle ambitious telecom challenges. You would be part of a growing tech team of 30+ people that builds a cutting-edge international VoIP network on which relies our SaaS customer interaction management solution. This solution all...innovative person by nature, you are eager to learn new technologies and to take up new challenges. You master most of the following technologies and methods : * VoIP protocols (SIP, RTP, WebRTC) * Analysis and investigation (logs, IP frames, pcap traces...) * Shell and Python scripting * Automation (ansible) * Monitoring (collectd, grafana, nagios) Bonus: knowledge of Oracle Acme Packet SBC, Asterisk, FreeSwitch...

I want a asterisk call base caller/callee filter solution on PHP which can filter incoming traffic and create whitelist and black list base on condition of repeated callee and non repeated callee with voipcall bandwidth optimizer..It will also work from 1ip to another ip.. Blacklists and whitelists of phone numbers/ Caller IDs on the basis of repeated numbers. Maintaining Call Gaps between calls to simulate HB Monitoring number/Caller ID length pattern while receiving calls Optimizing Bandwidth Efficiently

Install and configure LDAP in Docker on AWS and write schema and queries for our application The preferred LDAP server is ReOpenLDAP Detailed requirements for LDAP 1. Configure to use LMDB as its data store. 2. Configure for graceful shutdown, 3. Auto start on Docker launch, 4. High availability and horizontal scaling. 5. Write schema and queries for our application. 6. Integrate with MuleSoft using connector

I have trouble in getting users stored in OpenLDAP authenticated through squidguard 1.5.

I have a raspberry pi with a working asterisk 16 and chan_dongle I am looking for someone who can 1 capture dongle data from ami and update in db 2 assign static names to each port on my usb hub so that port 2 will always be port 2 3 insert dongle imei in a mysql table based on which ports the dongle is in 4. call a file if there has been an imei change on a usb port

Hi, I'm building a website with Gatsby. I need someone to fix an issue on the localization part, some pages work some don't. No agencies please. In your proposal, add three asterisk at the end.

We are migrating 10 Mailboxes and 5 shared mailboxes from a DoveCot Linux in house server to 365.(80GB Mail) The internal Linux server is also running LDAP Directory for the PCs and Samba File share Group drives (700GB). The office is using office 365 for Office applications only () and IMAP Connection to mailboxes. No Calendar or Contracts. I have Migrated the mailbox data (80GB) to the 365 mailboxes using a 3 month trial exchange licence and added the corporate domain, and added the corporate alias to each account. I have updated the CPanel DNS records apart from MX until I am ready to switch. For the Group drives I have created one sharepoint site, and copied 2 shared drives to the Sharepoint site, where the whole company has access. I have 9 other shared drives with

I want a new asterisk server built to replace our old version. i want someone who will build the server from scratch. Not just copy the sample files and fiddle with them. i require the following functionality. 1. Twenty Extensions. SIP Phone and Softphones etc 2. Conference bridge with IVR to Choose a room. Must support Video and Messages. 3. Short text message between extensions. 4. Video Calling between extensions. 5. IVR with Time of day manu for office and none office. Dial by name eyc 6. voicemail for all users. 7. outbound dial for two factor authentication. needs to call end user. ask question and receive response. (details from database ) 8. 4 Analogue Lines. Digium Card installed 9. 2 SIP trunks. Linked to group of extensions. 10. hunt groups. 11. Connection to Mic...

...developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different techologies by self learning. We are looking for attention to detail, communication and reliability. If you believe you are a good match for our team, please let us know your thoughts on the following challenge. If you had to create a voicebot integrated asterisk in 2 languages with a voicebot needs to do the following: 1st call : register user Ask what language would he like to speak. In the language of choice: Ask for birthday Ask for Name Ask for his address 2nd call : Tell the user what you know about him Hey {name} you are currently {age} in the right language. what would you like to know? User can ask: What's my name Wh...

Hello, the script already exits. There is a documentation. I need a full integration of the script in Node.js, one side FreePBX and the other Amazon Lex (or google speech). Possible upgrade with UniMRCP. here is the link

We need to implement the i...servers of MediaWiki. One server running CentOS 7 and MediaWiki 1.33.1 and another server running CentOS 8 with MediaWiki 1.34.0. Besides the configuration of this site, it's necessary to create a documentation to be reproduced in any MediaWiki site with the same versions of the same integration. I'll provide VPN access to both MediaWiki servers and an account on Active Directory to configure the LDAP bind. Before doing the job, I'll take a backup of database and of VM of MediaWiki. After finishing the job, I'll get the documentation provided by freelancer, and I'll restore the state before the job configuration and I'll follow the documentation to reproduce the configuration. The project is considered done and finished ...

Install LDAP in Docker on AWS and write schema The preferred LDAP server is ReOpenLDAP Detailed requirements for LDAP 1. Configure to use LMDB as its data store. 2. Configure for graceful shutdown, 3. Auto start on Docker launch, 4. High availability and horizontal scaling. 5. Configure and make ready for writing schema.

Hello, we created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho do...created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho doesn t appear any pop up Will be provided a vpn to access Asterisk pbx and admin l...

We are looking to own an open source(preferably) SIP media server and stream audio on it. This has to receive audio stream from asterisk. This has to start automatically when conference starts up on asterisk (our pbx). We have the PBX'es, just need that audio stream / internet radio that can be accessed by those who can't call in.

hello I hope you all are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. Budget is tight as well as time schedule but there is a great opportunity also for the maintanance phase.

...meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. Features needed Connection by phone (SIP trunk) This will be a SIP number that terminates on asterisk (freepbx) IVR which will request conference room number and pin based on the information (numeric only) provided by the script. This will require using asterisks API or .call file. Interface Main interface for users should be controlled by Username and password User should be able to create conferences, schedule by time, and so on. Create a free account on Zoom for live examples. Once a conference

I am looking who understand voip scanning . basically I am looking to install following program from start : please read and let me know if you can do this for me ?

Looking to rewrite existing Java servlet to work with Active Directory(LDAP)

we are interested in VOIP Asterisk PBX with ISP and IAX on Puppy Linux where Internet provder is blocking RTP/i/ports so need some encrption or tunneling also

Theory of operations: 1. Customer will dial a satellite nr 2. Asterisk will recognise the pattern and will add an additional string in front of the number

I am looking for some advice technical assist with an issue that we are having with our asterisk based PBX. We have been running an asterisk setup on our internal network for several years with almost no incident. We have two ISDN lines that connect through a B410 digium card to a wazo based asterisk box. Due to us now needing to work remotely we are running into issues. Everyone has their own physical SIP based phone at home that connect to our internal work network. This network has a public IP so there are no issues there. We have also set it up with the following IP rules: 5060:5160/udp ALLOW 10000:65535/udp ALLOW I have got the phones mostly working but I seem to be running into intermittent issues. These issues

I'm running a PBX (Private Branch Exchange), which should be accessable from the internet and from OpenVPN. The PBX is running in a datacenter, so there's no other NAT. OpenVPN's interface ist tun0 with Asterisk PBX is running on The "world" (the internet) is provided over eth0. OpenVPN is reachable from the internet on eth0 port UDP 443. The PBX should be reachable from the internet on port UDP 5060. Since some guys have NAT problems at home, the PBX should also be reachable from OpenVPN. I need someone, who designs the DNATing rules for me, perhaps a policy based ruleset is needed?

i want to implement webrtc on one of Customer development project I want to implement webrtc server software like Asterisk or FreePBX

We are looking for an asterisk / full stack developer who can help us build a custom dialer platform for call center using Asterisk as the backbone This project is a VoIP Platform for Agency Call Centers mainly for high volume lead call consumption across various types of health, insurance and mortgage products. Here is the basic outline of the project: This project will require the following: 500 outbound channels sip (spanability) Administrator GUI Agent Panel Phone list builder CRM - Client Manager with Followup Scheduler for Agents, commission tracking, retention Sales Reports Vendor Manager (Add, Modify, Delete Vendors) Lead List Dnc database and scrubbing tool HIPPA / PCI Compliant Statistics page .wav file uploader and player for outbound Email for call results .w...

I need someone to help setup an emergency call center using asterisk. The system should be able to handle 100 concurrent calls (server had been procured). We will start with 10 agents and they we would like to use a soft phone to connect the agents on their laptop. 1. Call Transfer, Hold, Forward 2. Call Recording and Playback 3. Configure IVR tree We will need the asterisk box configured with a local sip trunk provided by the telco. Soft recommendations welcome otherwise can purchase something like Please this is an urgent requirement and only have 3 days to deploy it. less

hello I hope you are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. I would like to hire you directly please contact me and disucss the project.

I am looking for some solution based on asterisk or freeswitch which provides me similar features as of multi tenant pbx like unlimited plan in please apply only if you can deliver, demo will be required.

I have Issabel CallCenter (elastix ) I need to add AMS (aswering machine detect) on this server Issabel working with Asterisk 11

Hi, I have a SIP Trunk from Saudi Telecom Company (STC) in Saudi Arabia and I want to configure inbound and outbound calls. The SIP trunk comes with static IP address and it's connected directly to the STC company. I need an expert who did this STC SIP trunk connection many times. I need him/her to send me the full configuration steps so I can do it. Note: I can not let you access the Elastix server so Please do not ask for that.

Hi arodrigue, I noticed your profile and would like to offer you my project. I want to make a phone verification system using asterisk or freeswitch. We can discuss any details over chat.

UI/UX Developer Web Developer Gamification Developer Asterisk Developer (VOIP, 911, PIDF-LO Stack, Dynamic Status) IOS Developer Droid Developer – NEED ASAP – current developer is leaving at the end of March Back-end Developer (need 2 of these)

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Hi. I need a push service for asterisk to support Android/iOS softphones.

I need someone who has multiple experience with all web programming and development. I have a lot of small task. So easy money for experience web developer. Skill Need: HTML, PHP, Laravel, Codignetor, Wordpress, Magento, Prestashop, etc. VOIP, Free PABX, ASTERISK, Linux, Centor, Ubontu Rest will discuss via chat. Write IDOUNDERSNAD then I can understand you read my project details. Note : My every task will be for 5-15 minuits for the one who has sufficient knowledge. AND payment per task will be 5-10. I have approx. 60 task. SO i you agree then bid other wise please dont bid. Thanks

We are looking for Asterisk developer to develop IP PBX and Dailer.

...to configure asterisk real-time to work with multiple tenants (e.g with tenant based contexts or dial plan aliases). Infrastructure exists in AWS for Database and auto scaling group of asterisk servers but the whole thing needs configuration. This is entirely from scratch. What we have ------------------------------------------- We have a fleet of raw, unconfigured asterisk servers in an auto-scaling group (ASG) in AWS. We also have a database instance in Amazon RDS - again, just the instance - no schema or configuration yet. We have all of the necessary networking infrastructure all configured in AWS with security groups, VPC etc... How it will work ------------------------------------------- Customers will have their own virtual PBX within this set up. We want ...

...blocked if the client who wishes to contract the service does not confirm. * From the administration page you can go with a click to the client panel (without authentication) * pop-up window of incoming calls with contact details, if any, to the extension. There are already commercial solutions at this point, but we need adaptation to our project. This is integration with Asterisk, although you do not need to know Asterisk or the SIP protocol, since the program will receive the events of incoming calls with the necessary parameters, such as tenant and caller id. You can choose the event-oriented language you know. If you wish, we provide the server for development. We know that the solution is developed and commercialized, but without access to the code related to multit...

Develop an inbound telephony script for asterisk What you will get - A machine with Asterisk installed, SIP information for the telephone connection, and MRCP servers installed for Speech to Text and Text to Speech. You will also get access to a web dialogueAPI and the input-output specifications of that API. What you need to do: Write a script that works according to the attached rules. Please read the attachment before bidding.