A2billing freepbx asterisk jobs
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...
I am seeking an experienced developer to assist with the deployment debugging of an Asterisk based CRM. The specific issue that needs to be resolved is integration problems with other systems. Skills and Experience Required: - Proficiency in Asterisk and CRM systems - Strong knowledge of integrating systems with databases, third-party software, and hardware - Experience in troubleshooting and resolving integration issues - Ability to work under pressure and meet tight deadlines, as this project is of high priority.
I am looking for a developer to develop a call forwarding pannel and pbx panel on freeswitch voip. The call forwarding panel and pbx panel should include advanced call forwarding with scheduling and should include integrated voicemail and call recording. I am not sure which VoIP provider to use and would appreciate advice in this regard. I ...appreciate advice in this regard. I am targeting completion within a month. The successful developer will be knowledgeable and experienced in using freeswitch VoIP in order to complete the project. If this project is completed to my satisfaction, I will consider having the developer complete additional projects with me. Looking forward to hearing from an interested and qualified candidate. No asterisk Candidate. Only Freeswitch other voip devel...
I have a VOIP PBX based on Freepbx with 2 extension . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is activated with the night and day option with code *280
I need asterisk to do the following: 1. asterisk Dial to callee 2. Callee pick up the call and the call is answered/ bridged 3. After X seconds, asterisk will inject DTMF tone 4. Caller will not be able to hear the DTMF.
Hi Aqs Y., I'm looking for a consultant to upgrade from freeswitch 1.10.6 to the latest on debian 10 (most probably also debian needs to be upgraded, to 11 or 12, as you prefer) It needs to work with asterisk 16.29.0 and also needs to support mod_expr, ceil and randomize
We are seeking an experienced freelancer to facilitate the seamless integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and ...integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the utmost security for our call traffic through Encrypted Media. Project Requirements: -Integration of WebRTC functionality into the Isaaabel platform. -Configuration of JsSIP npm package to establish connections with our Asterisk calls to SIP and PJSIP extensions through the WebRTC interface. -Implementation of robust security measures, including encrypted media, to safeguard call communications.
I am looking for a freelancer who can help fix the issue with the Vicidial inbound carrier stats for my existing business. The specific problem we are experiencing is missing data in the reporting. We need this issue to be resolved within one day. Skills and experience required for this project: - Expertise in Vicidial and inbound carrier stats - Strong troubleshooting and problem-solving skills - Experience with data analysis and reporting - Familiarity with call center operations - Ability to work efficiently and meet tight deadlines
I am in need of assistance with a clean up and addition of features to an Asterisk PBX system. Specifically, I'd like assistance with setting up an IVR menu which can be used to direct customer calls to the right departments. Additionally, I would need the system to be maintained regularly after installation of the features. I have a preferred method of communication through email and am eager to find an experienced freelancer for the task.
I need to modify the database CRD report in FreePBX. Specifically, I need to add more minutes upon disconnect call. I would like the modifications to be completed as soon as possible. I need to add all new fields to the database CRD report. Ideal skills and experience for this job include: - Strong knowledge of FreePBX and its database structure - Experience with database modification and customization - Proficiency in SQL - Attention to detail and ability to accurately add new fields to the report
Please don't bid if your bid request is more than 20 usd. We are looking for someone for installing Ubuntu Linux 22 and free pbx. Also configure apache for multiple domains and subdomain with latest php 8.1+. At the end provide a iso image that can be used to reinstall in any x64 machine.
I am looking for a freelancer to help me integrate a WebRTC based calling experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encry...experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement so it would be great if the freelancer can deliver the project quickly. Resume, we need to connect WebRTC extension with Issabel. I need that JsSIP npm package can connect with my asterisk server and make calls to SIP and PJSIP extensions. I need the documentation of the implementation for future re-ins...
I am looking for an experienced Asterisk developer to create a web interface to manage all of its features. Specifically, I need call routing and forwarding, an Interactive Voice Response (IVR) system, and call recording and monitoring capabilities. No specific requirements for the web interface are necessary and the freelancer will not be expected to provide post-development maintenance or support.
...skills - Attention to detail for accurate data entry and call tracking Project Description: We are looking for a freelancer who can assist us with our Vicidial short duration project. The desired duration of each call is less than 1 minute, and the purpose of the calls is sales. We require someone with experience in using Vicidial or similar call center software. This project can be done on asterisk of vicidial web interface as well. Itshould be just a code on the dialplan. The specific features for the dialer have not been specified by the client. Therefore, we are open to suggestions and recommendations from the freelancer. The ideal candidate should have a strong background in sales, with proficiency in sales techniques and strategies. They should be able to ha...
