A2billing freepbx asterisk jobs
I am looking for a freelancer who can integrate our Odoo community with our Asterisk-Issabel Phone Center. The specific feature that we want to integrate is the ability to be able to make calls straight from Odoo. We are open to suggestions for the preferred method of integration. The timeline for this project is immediate, with a deadline of 1-2 weeks. Ideal skills and experience for this job include: - Strong knowledge and experience with Odoo community and Asterisk-Issabel Phone Center - Proficiency in CRM integration - Familiarity with different integration methods, such as Direct API and Web Services
i want to implement a simple architecture. at this moment my asterisk server works fine with endpoints and it can initiate calls in both directions. now i want to use a proxy server in between endpoints and my server . so im using dsiprouter with domain name of and its installed successfully. now there is a problem. i connect my sip users to dsiprouter. and dsiprouter to asterisk as pass thru. when i call any number with sip endpoint there is no problem and the call reaches the dinstar gateway and dinstar call the sim number(ip2tel). but the opposite side of call when sim user call the gateway and gateway route the call through asterisk. when asterisk send request to dsiprouter the request cannot find the endpoint i tried (realm-outbound proxy and etc) but i di...
I am looking for a freelancer who can help me with the installation of VitalPBX 4 (based on asterisk) connected with a Twilio number. OS is Debian. Additionally, I require complete configuration of this addon: (it has a .sh script auto-installation) After all is done, YOU MUST GUIDE ME to make the first succesful call with the addon.
Project Description: I am looking for a skilled developer who can create a WhatsApp to SIP Gateway using Asterisk. The main function of this project is to enable call forwarding between WhatsApp and SIP. I require the gateway to be built on an open-source platform. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. Skills and Experience Needed: - Strong experience with Asterisk and VoIP systems - Proficiency in working with both WhatsApp and SIP protocols - Knowledge of call forwarding and routing techniques - Familiarity with open-source platforms for building gateways - Ability to troubleshoot and debug any issues that may arise during the development process
...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which...
I am looking for an expert freelancer to integrate SIP services with med...freelancer to integrate SIP services with mediasoup for my project. The purpose of this integration is to enhance the video calling capabilities on my application. Requirements for the project: - Experience: The freelancer should have expert level experience in SIP video integration with mediasoup This Project may require expertise in VoIP technologies along with WebRTC, Asterisk, kamailio etc. - Tools and Technologies: NodeJS is preferred, however to get the project integration we are open to suggestions and do not have any specific tools or technologies in mind. If you have the required expertise and can provide guidance on the best tools and technologies to use, please bid on this project.
I need to install a2billing + freepbx on my server. Create a SIP account and configure a trunk. It must be proven that it works. within 24 hours.
I have vicidial and asterisk installed, I need the following and the necessary configurations in vicidial for this to work. 1.- Create 17 trunks that are in the local network, [15 goip + 2 skyline gateway] 2.- Generate a way to call certain trunks together, easily, hopefully with just a few clicks: Example 1: call trunk 8, number 103 and play the dtmf tone of * Example 2: call trunk 9, number 103 and play the dtmf tone of * Example 3: call trunk 10, number 103 and play the dtmf tone of * Example 4: call trunk 16, number 103 and play the dtmf tone of * Example 5: call trunk 17, number 103 and play the dtmf tone of * Example 6: call all trunks 1, 2, 3, 4, 5, 6, 7, 11, 12, 13, 14 and 15 together, to number 103 and play the dtmf...
I am looking for a freelancer who can configure Asterisk freePBX with Linphone client for ZRTP encryption to enable secure VoIP communication. Here are the requirements for the project: - The primary purpose of the configuration is to enable post quantum encryption for secure VoIP communication. - The specific version of Asterisk and Linphone is not specified, any version will do. - There will be no requirement for ongoing maintenance or updates after the initial configuration. Ideal skills and experience for the job: - Strong knowledge and experience in configuring Asterisk freePBX and Linphone. - Familiarity with ZRTP encryption for secure VoIP communication. - Ability to work independently and deliver the project within the specified timeline. If y...
I am looking for a freelancer who can help me restore a FreePBX on a new Contabo VPS server. Here are the details of the project: Version of FreePBX: 14 - The old server has FreePBX version 14 installed. Backup of old FreePBX server: Yes - I have a backup of the old FreePBX server. Preferred VPS server: Contabo VPS - I prefer to set up the new FreePBX on a Contabo VPS server. Skills and Experience: - Experience with FreePBX restoration and migration. - Proficiency in working with Contabo VPS server. - Knowledge of FreePBX version 14. restore all configuration of old server If you have the necessary skills and experience, please bid on this project.
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between F...
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between F...
I am seeking a freelancer to assist me with setting up my issabel PBX system for incoming and outgoing call recording. Requirements: - Experience with issabel PBX system - Knowledge of Asterisk, FreePBX, and Elastix - Ability to configure the PBX system for call recording - Familiarity with telemarketing purposes for PBX setup Tasks: - Set up the issabel PBX system for incoming and outgoing call recording - agent call out with display random cli, if customer missed this call and return a call will back to particular agent. - Configure the necessary settings for seamless call record - Ensure compatibility with our existing hardware and software - Provide guidance and recommendations for optimal telemarketing functionality If you have the required skills and experience in ...
**Job Title:** Kamailio & Asterisk VoIP Solutions Architect **Job Description:** We are seeking an experienced Kamailio & Asterisk VoIP Solutions Architect to design and implement a comprehensive telecommunication solution that integrates WebRTC, SIP users, and robust VoIP services. The ideal candidate will have extensive knowledge in setting up and configuring multi-tenant VoIP systems, ensuring high availability, and developing secure and scalable APIs for system interactions. **Key Responsibilities:** - Design and deploy a multi-tenant VoIP solution using Kamailio for SIP routing and trunk configurations, and Asterisk for media processing. - Implement WebRTC functionality for browser-based communication, alongside traditional SIP user capabilities. - Con...
...freelancer who can help me integrate my VoIP Asterisk phone system with Google Sheets. I want to track caller ID and call details in real-time. I've attached an example of that is required, the phone system is Yeastar P-Series I will provide the Asterisk manager interface details Skills and experience needed: - Proficiency in Asterisk and VoIP systems - Experience with Google Sheets API - Knowledge of real-time data integration - Strong problem-solving skills to troubleshoot any potential issues Project requirements: - Connect the VoIP Asterisk phone system to Google Sheets - Set up a live CDR feed to track caller ID and call details - Ensure real-time updates of the data in Google Sheets Please note that there are 1-5 phone lines connected to the Vo...
I need a freelancer who can troubleshoot and repair my FREEPBX system as I am unable to make outbound calls. Although there are no specific error messages, there are unusual system behaviors. I have not made any recent changes or updates to the system. Skills and experience required for this project include: - Expertise in FREEPBX system troubleshooting and repair - Knowledge of VoIP protocols and configurations - Familiarity with Asterisk and Linux operating systems - Ability to diagnose and resolve issues with outbound calls - Strong problem-solving and communication skills The freelancer must be available to start working on the project today.
...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which t...
Necesito instalar freepbx + a2billing + 2 troncales en red local y 2 usuarios para llamar.
Linux / Asterisk server with 8 port sim gateway outbound call center. (partially inbound) agents with ip phone/gsm phones Agents must able to hold/CB/transfer call to supervisor and other functions with key buttons only. other all standard features. daily excel report for whatsapp pls quote.
Looking for a FreePBX Expert that can help configure my instance. Also I have this extension for my CRM which I need assistant with connecting to my instance and configuring.
...hardware. The requirements for this project are as follows: Operating System: - The VOIP server should be installed on a Linux operating system. VOIP Software: - The specific VOIP software that should be used for the PBX installation is Asterisk. Features and Functions: - The VOIP server should have the capability of call forwarding. - No other specific features or functions are required at this time. Ideal Skills and Experience: - Experience in installing and configuring VOIP servers. - Proficiency in working with Linux operating systems. - Familiarity with Asterisk VOIP software. Please provide your relevant experience and any certifications you may have in this area. Additionally, if you have any suggestions or recommendations for the hardware or software configura...
Develop a freePBX module with the following functionalities for utilizing my chan dongle to send and receive SMS: 1. Establish an Incoming SMS Portal 2. Enable SMS Sending 3. Facilitate USSD Sending 4. Provide Balance Checking 5. WEBRTC ( Call over web) 6. Support Dongle Setup and Configuration
This is the work that need to be done: 1. you will setup a Freepbx on a provided with any desk connection. 2. you will setup the SIP Trunk for incoming and out coming calls. (I provide you all the setting data) 3. you will setup the IVR. (simple IVR with different message for opening time and closing time) 4. you will setup 5 extensions 5. you will configure 3 Cisco Phones with the FreePBX, the phone need to be registered and tested for incoming calls, out coming calls, hold, transfer call. 6. you will setup the pfsense with freepbx rules 7. We together do a full test of the system to be sure that is all working.
I need to setup a new freepbx machine, 1 SIP TRUNK, 3 Cisco SIP Phone (You need to be specialist at setting this), 1 IVR.
I am looking for a freelancer who can help me set up FreePbx/Asterisk and provision 4 SIP phones for my VoIP phone system. Current Phone System Setup: VoIP phone system Software Installation: I already have the FreePbx and Asterisk software installed. SIP Phones: I am using Cisco SIP phones. 2xCP-6851-3PCC Phones 1xSPA-303 Phone 1xGigaset C530A Skills and Experience: - Experience with setting up FreePbx and Asterisk software - Knowledge of provisioning SIP phones, specifically Cisco phones - Familiarity with VoIP phone systems and configurations I NEED AN EXPERT IN CISCO PHONE.
Magnus Asterisk Inbound and Outbound DID Setup Skills and Experience Required: - Proficiency in Asterisk setup and configuration - Experience in setting up both inbound and outbound calling - Familiarity with DID providers and integration - Ability to customize Asterisk setup at a basic level Project Description: We are looking for a freelancer who can help us set up an Asterisk system with both inbound and outbound calling capabilities. We already have a DID provider in place and require basic customization for our Asterisk setup. Tasks: - Configure Asterisk for inbound and outbound calling - Integrate our existing DID provider with the Asterisk system - Customize the setup at a basic level to meet our requirements If you have experienc...
I am looking for a freelancer who can assist me with setting up a freepbx server. I already have all the necessary hardware and software in place. I need the freelancer to configure asterisk to meet my business needs and migrate my existing PBX to Alibaba Cloud. The ideal candidate should have experience in freepbx, asterisk, and Alibaba Cloud. The project should be completed within a week.
hi i have run voip example on esp32 lyraT kit and used local sip server(minisip) then it is working fine for call , but i have hosted the asterisk sip server on goolge cloud (the asterisk is working fine as i tested by calling using mobile apps. ) but when esp32 connects with this asterisk server whenever i call from mobile app upon pressing play button it says " no body is available to attend your call" .
Please help me connect JIO sip to my freepbx server... Ping ia working perfectly.. Only outgoing is not going to any number thiugh numbers are running and incoming is also not working..
I have a server with an FXO card, the scenario that i need to do it is this: 1.- i have extension 100 with a mobile phone in a follow me destination 2.- if the mobile phone it is not picked up in 9 or 10 seconds needs to send to voicemail or IVR or another destination (like...scenario that i need to do it is this: 1.- i have extension 100 with a mobile phone in a follow me destination 2.- if the mobile phone it is not picked up in 9 or 10 seconds needs to send to voicemail or IVR or another destination (like a ring group) the main problem is that i cannot make the asterisk to understand when the PSTN call exceeds that ring time because for asterisk the call it is answered so i need a help to setup correctly my box for make it to work please only experienced freepbx...
Hi I'm looking for an Asterisk AGI written in GO that is probably going to use this library: and which is called from the dialplan as: exten => 500,1,AGI(gotest,${myVar}) exten => 500,n,HangUp and is able to: * read the 'myVar' variable * read the 'agi_extension' * print to syslog and exit if some variables are missing * execute a saydigit(123) * execute the playback of a wav file * use get_data to get a digit and log it to syslog * set the callerid to 456 * execute a dial(SIP/789) with max ringing 60 seconds and return the ANSWEREDTIME and DIALSTATUS arrays * hangup max bid is 100 euros you must have your own Asterisk setup and GO environment and provide instructions on how to setup and build the code.
...WhatsApp number. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to the SIP backend like CALL SUCCESS(200 OK), BUSY(486 Busy Here), UNAVAILABLE(503 Service Unavailable), etc.... to try other rounds on the other SIP Switch. Functional flow 1) Calls from PBX/sip gateway will be sent to WhatsApp gateway 2) WhatsApp gateway converts the SIP to WhatsApp protocol 3) If it i...
I am looking for experienced devOps that can work on setting up Asterisk, implementing Vosk STT, and setting up TTS. I have a server in place for the project, so devOps with expertise in this required infrastructure is of utmost importance. The ideal candidate should have good skills in Asterisk, Vosk STT, and TTS in order to execute the project successfully and to my satisfaction. Only applications from expert level devOps will be accepted.
...this language and dialect. Example: French (Quebec) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the French language, the file uses English words, then you need to put "French (Paris), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your notes, whic...
Need to install and then configure so I can receive calls. Immediate work.
...language and dialect. Example: Japanese (Hachijō) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the Japanese language, the file uses English words, then you need to put "Japanese (Kyūshū), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your notes, w...
I have Job Listings, from Indeed, in Workbook, on SHEET 1. SHEET 2, i want to have/use, 3 Columns for FILTER the data in SHEET 1. COL_1=Job Title or Position, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER (add/approve/allow) from SHEET 1 to SHEET 3, final OUTPUT SHEET COL_2=Company or Business Name, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER OUT (remove/ignore/delete) from SHEET 1, to SHEET 3, final output SHEET. COL_3=Location, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER (add/approve/allow) from SHEET 1 to SHEET 3, final OUTPUT SHEET SHEET 3 - is the OUTPUT, from ALL DATA in SHEET 1, using the SHEET 2 Filters (3 columns with a FILTER/APPLY button), and SHEET 3 ends up with the res...
**Project Description:** **Overview:** We are seeking a skilled developer to integrate Asterisk PBX into our system for the purpose of answering incoming calls, transcribing audio to text, and converting text to speech. It is essential that these processes are performed offline, without reliance on external hosted solutions like Google. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in Asterisk, C programming, and offline audio processing. - Examples of previous projects or work that demonstrate your skills in these areas. - An overview of your approach to achieving the specified objectives. - Your proposed timeline for completing the integration. - Your pricing structure for t...
I am looking for a freelancer who can troubleshoot my Issabel (Asterisk) configuration with GoIP. Specifically, I am experiencing connection issues and there are error messages that need to be addressed. Ideal skills and experience for this job include: - Proficiency in Issabel (Asterisk) configuration - Experience with GoIP configuration - Knowledge of troubleshooting connection issues - Ability to address and resolve error messages. The successful candidate should solve any configuration issue with Goip to handle in-out connections
I'm looking for an experienced software development team to help build a custom Asterisk or Freeswitch speech-to-text system for my company. This system needs to have the ability to convert live speech and respond back by pressing DTMF digits. (Example: inbound call will play "press 3 to continue" ,at which point your software would press the digit 3 or whatever digit is in the initial spoken phrase. Then, the inbound call (assuming you pressed the right digit) will say the next phrase. Whatever is said in that next phrase would have to be speech2text converted and saved to db & HTTP POSTED to a remote url). I am looking for this project to use cepstral or some other free speech2text software - not looking for paid APIs) If you have experience with similar projec...
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Kamailio - Configure TLS certificates, will need to work with wildcard cert - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
I am looking for a freelancer who can help me fix the disk space size/partition on my FreePBX install. Here are the details of the project: Operating System: CentOS Preferred Method for Partitioning: Manual Partitioning There is this error: RedisException: MISCONF Redis is configured to save RDB snapshots, but is currently not able to persist on disk. Commands that may modify the data set are disabled. Please check Redis logs for details about the error. When checking the log file, it says: 13247:C 05 Oct 07:51:08.072 # Failed opening the RDB file (in server root dir /var/lib/redis) for saving: No space left on device 1347:M 05 Oct 07:51:08.230 # Background saving error Partitiion info is: nvme1n1 259:9 0 8G 0 disk └─nvme1n1p1 259:10 0 4G
I am looking for a freelancer who can help me integrate video-calling functi...video-call is to enable communication between two extensions on a Raspberry Pi using Node-red code. I already have a script with audio-calling, but i need to make it video-calling with VP8 codec. I need someone that know what he/she is doing...I can provide the script of audio-calling, to help him/her change it. I will pay 50 dollar. Skills and experience needed: - Proficiency in Node-red and Asterisk - Knowledge of video-calling protocols and codecs - Experience in integrating video calling functionality into existing systems Preferred libraries and tools: - The client is open to suggestions for libraries and tools to use in this project. Timeline: - The client expects the project to be completed with...
...via integration with the FreePBX platform. ** Primary Responsibilities: 1. Development: Develop a Flutter application with a clean and user-friendly UI for video calling. 2. Integration: Effectively integrate the application with FreePBX to ensure seamless video communication. 3. Sample Code: Provide well-documented sample code to demonstrate the integration and functionality of video calls with FreePBX. 4. Testing: Conduct rigorous testing to ensure the application’s stability, reliability, and performance. 5. Support: Offer technical support and troubleshooting assistance during the integration process into our existing application. ** Required Skills and Experience: - Proven experience as a Flutter developer. - Extensive knowledge and experience wi...
1st phase Currently, the active server is in 131 . Install and configure the new version on 130. After all done, change the IP from 130 to 131 (Change the IP from 131 to 130). And wait 1-2 days to see if any clients have complaints or no problem:). ( at least we have backup 131 old sever we make it 130 for backup) 2nd phase After 1-2 days good to go then we do upgrade new version and config to 130 (previously 131) After all is done, change the IP from 130 to 131 (Change the IP from 131 to 130 again). And wait 1-2 days to see if any clients have complained. 3rd phase Then we can do a load balance between 131 and 130. ( to test load balance ) 4th phase Step by Step installation and configuration video from 1st phase to 3rd phase
Build a web GUI for Asterisk 20 from scratch Requirements: - Programming language: C, Python, or any language that can fit the project. - Specific features: understanding asterisk - Desired timeline: Within 3 months We are looking to build a web GUI for Asterisk 20 from scratch. The GUI should have all asterisk features. We develop asterisk codes and config files by ourselves and need to build web GUI to allow users to manage asterisk features - The project should be completed within 3 months. Ideal Skills and Experience: - Strong proficiency in C, Python, or any language that can fit the project. - Experience with Asterisk and building web GUIs for telephony systems. - Ability to work within a specified timeline and deli...
...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which t...
