Asterisk elastixKazi

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 and see attached summary for what is done so far

i have server cloud and IP telephone i need install Asterisk free pbx to connect our getaway

A php script to manually add a phone number and Queue ID into FreePBX's Queue call back option to be called back at front of Queue. Run as a pipe from bash. The idea is for the FreePBX callback module to do all the work we just need to add the phone/queue number into their system. FreePBX Asterisk 16.6.2

I need help for asterisk goip long term , I have special dial plans that help to avoid sim blocking , I need to fine tune these dial plans .My budget is 50$ .

we have an asterisk module as a POC we need to create it for production enviorment.

We are looking for a VoIP systems engineer to help complete the development and testing of an internally developed web based calling application linked to Asterisk/Free PBX. - Experience with Asterisk and/or FreePBX - Experience with Apache, CentOS, DNS, hosted services, MySQL - Network design – Working with Firewalls, DNS, Load Balancers - Experience in software as service architectures (SaaS) - General telephony understanding - Understanding of VOIP platforms like Trixbox, Elastix, Freeswitch or FusionPBX - Configuring various VOIP Phones and iOS/Android Smart Phones - Knowledgeable in IP Telephony, unified communications, data networking, telecommunications, video technologies and Call Center - Experience w...

I need to fix a bug on yeastar tg1600 on the sms to email side. can you help me? is an asterisk appliance that stopped to send email when received sms 3 months ago I don't know why

Hi, I have a new server that has been apparently - partially-- built. The new server is to replace a hacked / rooted server. While attempting to terminate a call through the new server the call terminator is getting error code 503. Unfortunately the developer is missing not responding.....

I've done a lot with Asterisk and configured many Cisco phones to work with it but this one has me stumped. I'm looking for someone that has successfully configured one of these models to work on an Asterisk server. I'm not using any GUI like FreePBX. I have a custom-compiled Asterisk running on Ubuntu 18. Basically I'm looking for remote support to get this done. Thanks!

i need to integrate asterisk with php to make popup window when receive new call

i would need some statistics from our asterisk/freepbx server we need to read from AMI directly (with node.js server) i would need a statistics cronjob which run every minute and enters data gathered into a mysql table data that shall be fetched: calls in the queues (customers waiting) calls in progress (customers calling with agents) waiting time per caller in the queue Talking time per caller in the queue we have more queues so queue info must also be stored We have already installed 1)Node.js 2)pm2 already there runs a process which processes other information so the scheme is already setup- we just need an experts who extends the functionalty

Software to easily make Invoices with Open Source Magnus billing and/or FreeSide Billing, for Asterisk PBX. To be used with Ubuntu, LibreOffice. Details to be discussed.

vtiger crm installation, customerization and CTI asterisk integration in house by an expert having minimum experience of 5 years in thi s field

I need OAM software and initial Topex MultiAccess GSM Gateway installation and configuration. Following steps are requied: - instruction how to connect GSM Gateway with PC - installing OAM software (software must be provided, I don't have) - network configuration - setup SIP Trunk to local Asterisk server - configure outbound and inbound routing - testing

Hi everyone. I need a virtual landline phone number where people can call and speak to each other 1 on 1. In the attachment you will see a little example what we are planning. Thanks in advance. Feri

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

NEED SOMEONE TO MAKE ME A WEBSITE FROM SCRATCH ELASTIX WEBSITE

Create VOIP server on aws using Asterisk frameworks

we'd like to build Asterisk click to call API. Please check the attached PDF and let us know if you are able to deliver the task. Note: we need ASAP.

I am looking to develop a click to call API to be used with Asterisk. You must have good experience with Asterisk/AMI/ARI. All details attached in PDF. Please don't bid if you have never done this before.

We are looking for an asterisk expert, can help with a minor thing via anydesk. We must have transferred some data to our database with each call

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 somebody to review an existing freePBX / Asterisk installation. I want to connect two Trunks properly and use two Cisco Phones (CP8861 3PCC and CP8821) with it. One Trunk and one number is set up correct and works with Zoiper on mac. All other numbers (4) are set directly to on Cisco CP-8861 (working fine), but i want to move them to my freePBX Server (hosted on a vserver).

...protocol, and at least one common audio codec. Many service providers use the Session Initiation Protocol (SIP) standardized by the Internet Engineering Task Force (IETF). Skype, a popular service, uses proprietary protocols, and Google Talk leverages the Extensible Messaging and Presence Protocol (XMPP). Some softphones also support the Inter-Asterisk eXchange protocol (IAX), a protocol supported by the open-source software application Asterisk." Basic Fetures "A typical softphone has all standard telephony features (DND, Mute, DTMF, Flash, Hold, Transfer, call history, call outcome/disposition etc.) and often additional features typical for online messaging, such as user presence indication, video, wide-band audio. Softphones provide a variety of audio c...

I am currently sending SMS with AT commands in Text mode. In order to better support concatenated SMS, line breaks, and encodings, I want to switch to PDU mode. I need a PHP function that will generate PDU to be sent by Asterisk. Output need to be a JSON array with the PDU generated and part number, and total number of parts. Maxumum parts is 10. If more than 10 parts, error needs to be returned. If a mandatory parameter is missing or if parameter has wrong format, error needs to be returned. Parameters sent: *Destination number in international format (ie: 14152470402) *Message – SMS content Validity Period - in hours (if empty, Default is 72 hours) Status Report Request - 0 or 1 (if empty, Default is 1) Response expected: Status - ok (success) / ok (error with code) Numb...

I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

Needs integration asterisk server with odus. Ai

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 any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future:...

need a long terms sysadmin expert in: SYSADMIN Linux: -freeipa -freeradius -general linux debug -ODOO debugger -php installation knowledge -apache installation knowledge -CentOS expert -docker expert -asterisk expert -freepbx expert -routing expert ( we use mikrotik routerOS ) -apache / mysql / postgre SQL -Google Cloud Platform expert for managing application, instance, docker, db etc... SYSADMIN Windows: -AD expert -various problem solving like antivirus, printer etc... if not have full knowledge please dont bind an offer...

We would like to use the API provided by Mobile carrier to do the following... 1) Activate SIM Cards for Mobile Services directly from our service portal We will need this API incorporated in our A2Billing Server using agent portals you will need to know Asterisk programming and know about A2Billing Software. We are ONLY looking for developers who have done this type of API development! We will provide server and access needed for this project. We will provide detail documentations for API. We will provide Details Document of the API once we accept bid. or speak with you. We are NOT looking for any website to be develop, so please do not offer.

About us: Negombo Park has an Italian website. We're looking for an experienced translator to collect all the website text and translate it from ital...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...

About us: Negombo Park has an Italian website. We're looking for an experienced translator to collect all the website text and translate it from ital...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...

FreePBX (Asterisk) Freelancer in bangalore, India

...need a sip-bridge application which will work in android. 1. Calls will start from asterisk with sip protocol. 2. The sip-bridge application which will work in android will take the coming sip requests and make the calls via skype/viber/bip. Calls will be transfer to the mobile number, not his skype/viber/bip (to the called person's mobile number). The info about which number is gonna be called will be in sip request which starts from asterisk. person who originates the call via asterisk will make a phone call with the person being called via his mobile number with the help of sip-bridge application and skype/viber/bip that sip-bridge application; all the sip signalisation messages between asterisk and skype/viber/bip should be exactly transfer end to...

ive issabel asterisk 11 and vtiger 7 installed and configured i've even configure Vtiger connector.. nowi just need you to configure the dialplan so the users will start getting the pop on on vtiger Inbound +Outbound (Click 2 Dial or Click 2 Call) these 2 should work

I want to create a kamailio dispatcher but does a lookup in dB to locate the asterisk server which accounted is located on.

...or maybe you have scripts/modules that already working with vtiger - DONT ASK ME ABOUT modules - i dont know, you need find a way or possibly you have already those in your vtiger voip installations, use github, google, 3rd party voip free etc modules for vtiger. I need in VTIGER - voip features: 1. inboud,outbout call recording, playback button. 2. inbound, outboud WebRTC 3. you can install asterisk or freepbx 4. log all installations and commands you will run on server. 5. inboud,outboud if not in DB suggest save as a new client. 6. Admin can listen all agents inboud/outboud recordings. 7. Agent can listen only own inboud/outboud recordings. 8. Suggestions, advices, ideas - how to make everything better. You will get access to: 1. Ubuntu 18.04LTS + AJENTI panel + Vtiger 7...

In a scenario where there is: - A Asterisk VoIP server (provided to you) - A phone extension connected to the Asterisk server (provided to you) You should build an Android APP that connects to the Asterisk Server and is able to do the following: A) Receiving a GSM Call and Making a VoIP CALL 1) Automatically answer phone calls 2) Take GSM Audio call and create a voip IAX2 protocol call to a voip server 3) The app should receive the audio from the GSM call, encode it and send to the voip server 4) The app should receive the audio from the VOIP call, encode it and send to the GSM call B) Receiving a VoIP Call and Making a GSM CALL - The APP should be listening to a "mqtt" topic on a server and receive instruction on what to do. - If the inst...

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

I have a need to test an installation of elastix or freepbx on a hosted server (Hostinger Malaysia) Its a test site at this stage to see if it will meet my clients requirements

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.

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...