Openser asterisk loadbalancing jobs

https://www.freelancer.com/projects/asterisk-pbx/Twixtel-export-all-datas-including-22224454/proposals

I have a asterisk server maintained by a freelancer, he is not responding from a few days ,I need to make changes, dial plan have all the required settings just it is disabled , I need some one to help me enable and disable dial plan , my budget is 10$ , you do not have to write anything just guide how to disable and enable dial plan.

Unable to play music file in FreePBX error "Unable to find an intermediary converter for /home/asterisk" And FXO Tele Routing & FreePBX Inbound Configuration need to be setup

these are the details We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19

Let me describe i...Numbers (fields, Name,Number,Status) and dial the number thrugh Asterisik server (with Originate) and the update the field status with with the sip response () for the call P.S. SIP Responses are not ultimat, it can either be channel statuses BUSY, NO ANSWER , INEXISTENT The varibles needed should be : How many seconds to try the call Asterisk Extension to be used in the test Custom Caller_ID setup filed (the option to put a list that can be used random is a plus) More details can be discussed if needed , as the uper is a generic idea Libraries that are to be used: AdminLTE + Datatables Please dont bid without reading it , its a quite simple project witch will follow a lots more like that if this goes well

We are looking to implement routing across 1 or more SIP trunks on asterisk based upon the percentage of traffic. for example we would like to be able to define an extension and route traffic on the following rule. 80% of traffic to SIPTRUNK_A 20% of traffic to SIPTRUNK_B These are just examples and the list could be longer with variable parameters.. we would loike the solution to look something like this : ;exten => _71.,2,Dial 80% (SIP/02076462185@) ;exten => _71.,2,Dial 20% (SIP/${EXTEN}@)

Setup Asterisks connect sip. Task: asterisk calls with sip to mobile number if number is Working hangup and save mobile in text file, if number doesn't work don't save. That's all.

I need an experienced person who have min 5-6 year experience in MYSQL Database and also have expertise in load balancing and also have expertise to handle the million of data at a time. Experience 5-6 years location : jaipur Please don't waste your and my time if you don't have experience in relevant profile

Hi, We have a Panasonic TDE phone system that has regular lines in it as well as Pansonic phones. We Added a SIP card to it. We have an asterisk pbx running freepbx. We want the two devices linked to eachother so that they can call eachothers extenions and that outbound routing from the asterisk will use the panasonics POTS lines. You will be responsible to program the routing on both sides. To ensure you read the entire details, mention the word "house". Thank you

This app for both iPhone and Android will allow cellphones to access the audio contents on our Asterisk IVR system. Once the phone is connected to IVR, the phone will allow user to listen to different menu selection by pressing the digits on their phone. This app is a private app for use of myself and selected customer; and free for my customers to install and use. The app should have configuration settings that includes Host, AccountID and Passcode. Once the phone is connected to the host, the caller will hear first an ‘entry’ message and then a ‘menu’. The menu will include up to 8 choices where the caller can select their desired selection using their cellphone keypad. The audio should be two ways as there is a choice to move the phone to the owners offi...

For freelancers specialized in Call Center software. To fix a non standard PBX: 1. Asterisk is very slow to start or doesn't start at all. 2. Agent Operator Panel must check extensions. 3. Install a new panel to see agent activity. 4. Reorganize our Notepad to capture more information. --- Additional documentation and information available.

Requirements: - place of residence - Ukraine; - desirable - experience in set...important: be accessible by phone, e-mail, skype from 9 to 19 on working days every day the ability to allocate 2-4 hours each day for work on projects; - quick response to inquiries for customers (within 24 hours). We offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers; - interesting projects that will allow you to grow professionally quickly; - you can read about us here:

My company is system integrator that provide IT solutions to my customer. I have Elastix IP PBX ver. 2.5. I need solution (script or something) how to send the call recording files (located in /var/spool/asterisk/monitor/year) from Elastix to external server. The sending file by scheduling. Thank you.

We need an asterisk IVR that run on raspberry Pi 3+ and have some particular features: - Integration with REST API - Use Google Cloud Text-To-Speech for reading data to callers - Create VOIP Softphone to integrate on Angular 4 APP Verifiable experience on asterisk is required. If you're interested, please send me a message and I'll give you more detail.

Channel Object to contain all columns that would display from `core show channels verbose` entered in the asterisk console: { Channel, Context, Extension, Prio, State, Application, Data, CallerID, Duration, Accountcode, PeerAccount, BridgeID, } Main routine needs to pull all active channel info, and store it in some type of array for parsing. Java App must also include: Function to terminate an active channel by ID Function to terminate an active Bridge by Bridge ID Function code to reload asterisk config. It's really just a basic framework that is needed.

Need to create High level definition document for an enterprise which uses open source VoIP system and integrated to the mailbox of users using Asterisk IP-PBX

I owned the dating app where having some problem with background calling isn't working. The person should having knowledge of linphone SDK using our own build Asterisk server

We would like to setup a PBX(preferred: Issabel) in our environment and need some help for the installation, setup and migration of our current installation. We want a Docker image.

i need an expert for asterisk , vicidial ivr outbound . i have PU when i put 8373 route code

You`ll be creating a Bot on Python that shall automate the described procedure below for each Incoming Dialed number into a FreePBX/Asterisk softswitch. Call Flow: 1. Customer sends INVITE with Dialed number --> Asterisk AGI will then execute the Bot automating request (can be Flask App through a Curl call): 2. Python bot will send a POST request to a website with the Dialed number and 2 other values that I will describe in detail if interested in this project. (no login required) and fetch the result to either aprove or reject the call. 3. If number dialed got an specific result (out of 2 possible results) we tell the Bot to understand that specific result as a good number, therefore call must be allowed to continue through dialplan context. A few notes to have in consi...

I need an expert asterisk , vicidial about ivr outbound .

I need an expert for asterisk , vicidial , for my ivr outbound .

We need installation of Vicibox9 in proxmox, with 7 nodes in total 1 is web and database and rest 6 are dialers with asterisk 13 of vicibox, all boxes should be SHH seperately from outside all boxes should be port forwarded to webpage of vicibox all dialers communicate with public network with DB server IP address only

We need installation of Vicibox9 in proxmox, with 7 nodes in total 1 is web and database and rest 6 are dialers with asterisk 13 of vicibox, all boxes should be SHH seperately from outside all boxes should be port forwarded to webpage of vicibox all dialers communicate with public network with DB server IP address only

Hello we are looking for a team to implement WebRTC Plugin for Chrome to connect with our server and be able to use Call, SMS and video features, see call history etc

The Fourth Estate Public Benefit Corporation is a civil society organization with a mission to democratize the news for the public benefit. --- We are looking for a freelancer to setup, configure and secure an Asterisk server. We will be using XiVO, it's an Asterisk front end. The OS will likely be Debian. Documentation for XiVO is at: The system will need to be configured and connected to Twilio, our SIP provider and connected to two extensions that will be setup for soft-phones. The server will need to be secured. ---- The winning Freelancer could expect ongoing work to administer and maintain this installation

We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document

i need to integrate asterisk with php throw websocket

...flow and enhancements in our current implementation . We can roughly divide these functionalities into 3 parts: A. Webrtc Audio - Work required 1. Establish the webrtc currently implemented in view of scalability, robustness and voice quality and its integration with our application (we'll do the .net part but need help in the asterisk part of it). For this, as of no, we are just using one extension and there's no way to know who is making the call, so changes in asterisk context may be needed, to capture and keep the agent id and the no to dial from . Also, group audio conferencing and transferring the call to an app flow will be required 2. Configure webrtc for incoming calls too 3. Invoke the call pop up from the ivr and connect the caller to an agent 4. Enab...

I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer ph...someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not on Asterisk. 2. Instruction may instruct using CLI or exactly where and how to modify FreePBX files 3. Logic of FreePBX related with these 2 tasks 4. Project will be considered as completed only after instruction (PoC) will be tested on FreePBX and con...

I want to be able to stream music for agents, from our PBX, to replace MoH files, played by Asterisk. The music will be for entertainment and make sure that agents are connected to the PBX. There may be different ways to accomplish this. Please also see attachment.

I installed Vicibox9 on 5 different servers, 3 Dell Poweredge 2950 Servers, 2 i5 Desktops. 1 Dell keeps getting time sync errors even though the time is fine, 2nd dell the asterisk keeps crashing every couple of mins, third dell the time just went blank and wont sync at all, the i5 desktops (Clustered) works for awhile then also gets time sync errors, then I need to reboot them after which I sync the time manually with yast then they come back online again. We need one solid working vicidial server working. Dell Poweredge 2950 Servers All: 2 X Intel(R) Xeon(R) CPU E5405 @ 2.00GHz Quad Core 8GB Ram Raid HDDs Dell1,2 HDD 1TB Raid 5 Dell3 HDD 160GB Raid 1+0 Internet Connection 2 X 10Mb Fibre lines

Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.

Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will not be considered.

...have no parameters and one will have the name, password, and the amount of money parameters. (2 points) 12. The password secret encryption algorithm is sequenced below: (10 points) 1. The first two characters will be moved to the end of the string of characters. 2. A random number greater than 0 and less than 10 will be inserted between the 2 and third characters in the string 3. An asterisk ( * ) will be place after the 7th character 4. The first character of the Players’ name will be the last character in the password. 13. The password is stored and displayed as encrypted. The Player class will also have a method to display the decrypted password. (10 points) 14. Add and use the Console class from chapter 7 or a modified version to validate all user input ...

Hello, we need asterisk AMI script (syntax) for yeastar PBX we need functions must work via triggering AMI commands described and tested : 1. hangup 2. mute 3. attended transfer 4. hold please only serious freelancers with experience. Please be aware that Yeastar PBX has limited manager commands

...Trying to update: Unsupported Version of Asterisk, You need at least 11.11.0 you have 11.8.1 Running Amportal: amportal a ma refreshsignatures Fetching FreePBX settings with gen_amp_conf.php.. /usr/local/sbin/amportal: line 52: export: `®': not a valid identifier /var/lib/asterisk/bin/freepbx_engine: line 100: export: `®': not a valid identifier Getting Data from Online Server...Done Checking Signatures of Modules... Checking announcement...Signature Invalid Refreshing announcement Downloading 42672 of 42672 (100%) The following error(s) occured: - File Integrity failed for /var/www/html/admin/modules/_cache/ - aborting (GPG Verify File check failed) Trying to update the key: [root@hfp ~]# sudo -u asterisk gpg --refresh-ke...

I have an Asterisk-FreePBX System with multiple disks that needs some fixes. If You are a specialist in this field, lets talk. --- This is not for people who think all the answers are on the Internet ! This is for experienced specialists. Requirements: Asterisk, FreePBX, SSL Certificates (Letsencrypte), Apache, multiple disks on system, web dev, PHP, etc. Will divide into milestones.

...following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real Time Control Protocol. (3) Programming Languages: (3)(1) C/C++. (3)(2) Java. (3)(3) Swift/Cocoa/Cocoa Touch. (3)(4) Linux Debian Application Programming Interface. (3)(5) Android APP Development. (3)(6) iOS APP Development. (3)(7) HTML & JavaScript & MySQL & PHP. (4) OpenSource Softwares: (4)(1) GnuGK. (4)(2) Free SIP APP's( Android, iOS ) Source Codes. (4)(3) OpenSER. (4)(4) Asterisk. (4)(5) Speech To Text. (4)(6) WordPress or Any Other CMS Software....

Hello, I have an asterisk PBX vers 11.22.0 . I am using a Polycom sound point IP 650. All works fine except for the transfer button. The transfer on polycom use SIP REFER to transfer the call. This is not working. Need help from anyone who know about the subject. Please respond to this project with "What up Dingo" at the beginning of your message so that I know you have read.

Hello I have Asterisk dialer and I need to set up speech to text transcription (ONLY) I use to use to use IBM watson api for this, but it has become too pricey. it is 1 Min length audio of ivr recordings each. But total millions of files. every 2-3 months 7 million 1MB, 1 Min audio files.

Hi We are looking for a freelancer experienced in Asterisk. Current developer works at another job, so you will work with me for a long term if you want. hourly rate is 25~35. 40 hours per week Thanks Anthony

This is an on-going project with various tasks managing asterisk Please apply only if you have experience in this.

I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer ph...someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not on Asterisk. 2. Instruction may instruct using CLI or exactly where and how to modify FreePBX files 3. Logic of FreePBX related with these 2 tasks 4. Project will be considered as completed only after instruction (PoC) will be tested on FreePBX and con...

We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a public IP (direct). Endpoints are Acrobits sofpthone users (android/iOS) connecting via WAN. there is nothing in between. All end-users use TLS+SRTP. and Acrobits Push. So ping is huge sometimes. I suppose qualify option may be the cause here. Official FreePBX forum treads ignore the issue and ask to order their paid support. As the issue is in the production system, there is no place for exp...

Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database

Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall VPN Client

Looking for Asterisk, FreePBX with WebRTC specialist to verify and fix existing PBX. This is no place for amateurs !

I have a running asterisk PBX - i will to rebuild new one with asterisk. i use Asterisk API, Databases.

Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action