Iax asteriskcông việc

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 ...

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

We need to develop a SIP to Viber/Whatsapp gateway. The gatewa...versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if the number is online, the call will terminated on Viber/whatsapp!. 4) if the number is not used in VIber it sends 503 error and rerouted to o...

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

...versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/bip gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if the number is online, the call will terminated on Viber/bip!. 4) if the number is not used in VIber it sends 503 error and rerouted to other gateways . Exactly the work you did before but this time sip to viber...

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

Original work only, No copying content on the internet. Example of Coloring pages - Consider complexity that a 3rd grader can handle. Ping me if you have more questions, What is required ? 1) 10 Original Coloring Pages - In SVG or AI Format. - A4 Size. 2) Above files as PDFs - A4 Size. 3) Once I see good work, Its going to continue long term. I am still giving artists orders to execute from my last gig. 4) Content should be returned in 3-4 days. I would be sharing the particular type of coloring pages to draw. Examples are, Coloring pages of flowers, jeep, car and so on. ( google them and you can see samples )

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...

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