Freeswitch dialplan Jobs
Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated). I already have a small but buggy "solution" that needs to be tweaked. We keep the CID info in freeswitch hash database and collect the data when the parked call is retreived.
Hi I'm looking for a Kamailio / Opensips platform for routing and cdrs. I'm currently using freeswitch (with no transcoding) but i think we can save lots of cpu with a 'SER derivate'. If you can help me, let's talk :)
Hi I'm looking for a Kamailio / Opensips platform for routing and cdrs. I'm currently using freeswitch (with no transcoding) but i think we can save lots of cpu with a 'SER derivate'. If you can help me, let's talk :)
I am looking for a freelancer who can help me integrate Google+AWS+MicosoftCloud Speech-to-Text and Text-to-Speech services into my freeswitch project. The ideal candidate should have experience with freeswitch and be familiar with the setup and configuration of these services. Requirements: - Experience with freeswitch - Knowledge of Native Speech-to-Text and Text-to-Speech services in freeswitch - Knowledge of Google Speech-to-Text and Text-to-Speech services - Knowledge of AWS Speech-to-Text and Text-to-Speech services - Knowledge of MicrosoftCloud Speech-to-Text and Text-to-Speech services - Ability to assist with setup and configuration - Able to complete the project within 2days
...looking for a Freeswitch expert who can help me set up a Freeswitch cluster on my existing servers. The ideal candidate should have experience in configuring Freeswitch for optimal performance and troubleshooting and fixing any Freeswitch issues that may arise. Specifically, I need assistance with: - Setting up a Freeswitch cluster on my existing servers - Configuring Freeswitch for optimal performance - Troubleshooting and fixing any Freeswitch issues that may occur Requirements: - Strong knowledge and experience with Freeswitch - Experience in setting up and configuring Freeswitch clusters - Proficient in troubleshooting and fixing Freeswitch issues - Ability to work within a tight deadline If you have the neces...
Project Description: I am looking for a freelancer who can develop a Freeswitch software with specific functionality in IVR (Interactive Voice Response). The software should be programmed using Python or what ever you suggest and needs to be completed within a week. Requirements: - Develop a Freeswitch software with IVR functionality - Program the softwareusing Python - Complete the project within a week Ideal Skills and Experience: - Strong programming skills in Python - Experience with Freeswitch and IVR development - Knowledge of telephony systems and call routing If you have the required skills and can deliver the project within the specified timeframe, please submit your proposal.
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.
...destination's 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 p...
Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated)
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...
Require Full stack expert freeswitch developer for development of call dialer api. Experience in Freeswitch is MUST
I am looking for a developer to develop a...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 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 ot...
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
Hi. Our backend is node.js and frontend is KMM. We need a setup for voice call by freeswitch. If you experience of this , i love to talk about the task. it's for application call between user (one to one)
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Require expert with strong knowledge in Voip, Signalwire, Bring-your-own-carrier. -Setup IP Trunks in Freeswitch to be routed according to own carrier trunk. -Manage VOIP Inbound Termination, Outbound Termination through own carrier -Must know Rest API, Integration, Relay etc
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 handle high call volumes efficiently, ensuring a smooth and effective sales proc...
I am seeking a freelancer with experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + module rules stored in Mysql tables + Fresswitch for CDRs + PHPMyAdmin server 2 : kamilio + LCR module + address module + module rules stored in Mysql tables + Fresswitch for CDRs + PHPMyAdmin I am looking for someone who can get this done today. The servers are already built. Need help fixing and troubling shotting configs. Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
I want to add a VoIP client into our app to make and receive phone calls. Looking for iOS and Android developer To be able to send and receive SIP calls to our Freeswitch server I want to use something that doesn’t tie us to a CPAAS platform that requires us to buy phone numbers from them. Since we have our own VoIP system .So something like pjsip -No need for registration- -no need for incoming calls- i just care about outbound calls for now Please note about your experience of PJSIP SDK
We are going to hourly project and work on Kamailio and freeswitch configuration according to client needs.
I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables + Fresswitch for CDRs server 1 : kamilio + LCR module + LCR module stored in Mysql tables + Fresswitch for CDRs I am looking for someone who can get this done today. The servers are already built. Need help fixing and troubling shotting configs. Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
PLEASE NOTED MY BUDGET IS 90$ FIXED. I NEED IT DONE 2 TO 3 DAYS. I WILL CREATE MILESTONE 40$ NOW. WHEN ITS DONE 100% I WILL CREATE REMAINING 55$..AND RELEASE BOTH ON SAME YOU NOT AGREE PLEASE DONT BID.. I ALREADY LOST 50$ FOR THIS TASK AND DID NOTHING. HI I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables server 1 : kamilio + LCR module + LCR module stored in Mysql tables Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables + Fresswitch for CDRs server 1 : kamilio + LCR module + LCR module stored in Mysql tables + Fresswitch for CDRs Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
PLEASE NOTED MY BUDGET IS 90$ FIXED. I NEED IT DONE 2 TO 3 DAYS. I WILL CREATE MILESTONE 40$ NOW. WHEN ITS DONE 100% I WILL CREATE REMAINING 55$..AND RELEASE BOTH ON SAME YOU NOT AGREE PLEASE DONT BID.. I ALREADY LOST 50$ FOR THIS TASK AND DID NOTHING. HI I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables server 1 : kamilio + LCR module + LCR module stored in Mysql tables Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
PLEASE NOTED MY BUDGET IS 90$ FIXED. I NEED IT DONE 2 TO 3 DAYS. I WILL CREATE MILESTONE 40$ NOW. WHEN ITS DONE 100% I WILL CREATE REMAINING 55$..AND RELEASE BOTH ON SAME YOU NOT AGREE PLEASE DONT BID.. I ALREADY LOST 50$ FOR THIS TASK AND DID NOTHING. HI I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables server 1 : kamilio + LCR module + LCR module stored in Mysql tables Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
We request the customization of the linphone app for the Android, iOS, and Windows platforms, incorporating our logo and company informatio...receive SMS text messages through the ASTPP PBX server Implementation of secure communications through zRTP, TLS, and SRTP Support for Bluetooth headset devices Language options: English, Chinese Integration with ASTPP PBX server to check balance Integration with ASTPP user accounts Message Archive Management (MAM) functionality Implementation of PUSH Notification from the ASTPP PBX server (FreeSwitch) Customization of the app with our logo and company information Provision of source codes. Kindly consider these requirements for the customized linephone app for the Android, iOS, and Windows platforms, and integration with our ASTPP P...
...in project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, here is
...in project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, here is
...destination's 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 p...
I am looking for a freelancer who has experience in Kamailio, Fresswitch, Mysql, and Linux operating systems. I need Kamailio installed on two servers server 1 : kamilio + dailplan module + dailplan module stored in Mysql tables server 1 : kamilio + LCR module + LCR module stored in Mysql tables Please respond with " I know Kamilio" at the beginning of your message so that I know you have read.
I am looking...the freelancer to determine the best configuration. Regarding the authentication method for the SIP server/client, I am not sure if a secure authentication method is required. Further clarification on this matter would be appreciated. Ideal Skills and Experience: - Experience with configuring Zenitel tcis3 system - Familiarity with Windows-based SIP server/client software, such as Asterisk, FreeSWITCH, or 3CX - Strong knowledge of SIP protocols and configurations - Ability to provide recommendations and suggestions based on project requirements The systems will be for a internal/private network. So should not be reliable on a internet connection. So no cloud solutions. This is a basic setup so tcis3 can call windows pc and Windows pc can call tcis3. Nothing els...
Project Description: features to implement in Asteris...handle call traffic more efficiently to avoid getting GSM cards blocked, (Issabel installed and Skyline 32/132 GSM Gateway available, and current on use platform where VOIP to GSM working perfectly ) Skills Required: Incorporates all the mentioned features Call Throttling Time-of-Day Routing Gosub and Subroutine Traffic Shaping and Qos Load Balancing Call Queues Limit Concurrent Calls Custom Dialplan Logic Project Requirements: - Assistance is needed in setting up a custom dial plan Ideal Freelancer: - Has experience in working with Issabel, Asterisk, and Skyline GSM Gateway - Familiarity with VoIP and GSM Gateways, and their configuration and management Note: Please include your relevant experience and expertise in your...
ng-voice, a trailblazer in the realm of open-source and VoIP technology, is currently seeking experienced freelance software developers to participate in various cutting-edge projects. Our cloud-native VoLTE/IMS solutions allow mobile network operators to ...execution. Required Skills and Experience: Strong interest in and understanding of telephony. Proficiency in C programming language Experience with signaling standards such as SIP, DIAMETER is a plus. Fluency in English; reading 3GPP specs should be a part of your skillset. Proven track record in adhering to deadlines and keeping projects on track Our Technology Stack Includes: Kamailio, RTPEngine, FreeSwitch Redis, MySQL, and RabbitMQ Docker; experience with Docker Swarm and Kubernetes is beneficial SonarCloud, BugSnag SIP, RTP...
...needs some integration into Asterisk software, I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, here is
We are looking to hire an experienced Full Stack Developer (7-8 years of experience in Node.js with Freeswitch on an immediate basis to help us keep growing. This position is full-time and can be done remotely. Requirements Strong organizational and project management skills. Proficiency with fundamental front-end languages such as HTML, CSS, and JavaScript or React.js Proficiency with server-side languages such as Node.js Experience working with Freeswitch Familiarity with database technology such as MySQL, and/or MongoDB. Excellent verbal communication skills. Good problem-solving skills. Attention to detail.
...needs some integration into Asterisk software, I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, here is
Project Description: Asterisk PBX DialPlan Customization I am looking for someone with experience in Asterisk DialPlan customization to help me customize my existing DialPlan. The ideal candidate should have knowledge and experience in Asterisk DialPlan scripting, as well as the ability to understand and modify existing DialPlan code. Skills and Experience: - Strong understanding of Asterisk DialPlan scripting - Experience in customizing existing DialPlans - Ability to troubleshoot and debug existing DialPlans - Familiarity with Asterisk configuration and setup - Knowledge of VoIP technologies and protocols This project does not require assistance with other tasks besides the DialPlan customization, and there is no need for assistance wit...
Project Details: We are looking for someone who can assist us with forwarding the audio from the Asterisk dialplan to a server in the backend for live audio transcription of every call. Our team will handle the configuration of the Asterisk server itself, so we mainly need your consulting and "know-how" skills to ensure a smooth connection and provide guidance. Additionally, having a sample server showcasing the connection would be beneficial. If you have experience working with Asterisk and possess the knowledge to facilitate this audio forwarding and transcription setup, I would be very interested in discussing the project further with you. Alternatively, if you know someone in your network who has the necessary skills and might be interested in this opportunity. Pleas...
...a HTTP GET will be used to retrieve the contents of the provided url. The value written to the function specifies the destination file of the cURL'd resource. Example: Retrieving a file 1 exten => s,1,Set(CURL(http://localhost:8088/static/)=/var/spool/asterisk/tmp/)) Note Icon If live_dangerously in is set to no, this function can only be written to from the dialplan, and not directly from external protocols. Read operations are unaffected. Syntax CURL(url,post-data) Arguments 1. url - The full URL for the resource to retrieve. 2. post-data - Read Only If specified, an HTTP POST will be performed with the content of post-data, instead of an HTTP GET (default). See Also 1. Asterisk 16 Function_CURLOPT ...
...providing an API-like interface to queue RVM drops and to be able to push CDRs based on webhook settings set on JSON request body. It should handle an audio file per request and be able to have multi-carrier support. It could keep a database for CDRs, configurations and eventually requests log for auditing. As this is for high volume, it should be developed on a compiled language like Go and use Freeswitch dialer for scaling communications stack (programming language and SIP stack could be discussed). It is required extensive experience working with Ringless Voicemails on a high performance scenario and with verified high drop ratio, as well as experience working with SIP communications. This is a backend project, no UI is required and it will work as a module of a bigger ...
We need a skilled freeswitch or asterisk developer to help develop an API interface capable of initiating phone calls on freeswitch (similar to Twilio APIs for voice calls). Sample use cases are to embed the API in a CRM or webservice to enable to user click a button on an external web application, and initiate a call on the freeswitch machine and the call is then connected to a softphone or webrtc endpoint for audio.
...looking for a Kamailio Developer! Our in-house Kamailio developer is retiring. We would love to find someone who is a real Kamailio expert that loves creating/developing outside-of-the-box solutions. We want to find someone who has experience with custom modules, and who has a fantastic fine-tuning skill set. All our Kamailio are extremely high CPS it would be ideal if you have experience in Freeswitch & Postgress as well. This is not a one-project job and done, we would love to find someone for the long term and become our "Go to Guy" and have someone get familiarized with our network/systems. The first task we have is for someone to review one of our new configurations and see where we can fine-tune it, also want to change how Kamailio communicates to the cus...
...looking for a Kamailio Developer! Our in-house Kamailio developer is retiring. We would love to find someone who is a real Kamailio expert that loves creating/developing outside-of-the-box solutions. We want to find someone who has experience with custom modules, and who has a fantastic fine-tuning skill set. All our Kamailio are extremely high CPS it would be ideal if you have experience in Freeswitch & Postgress as well. This is not a one-project job and done, we would love to find someone for the long term and become our "Go to Guy" and have someone get familiarized with our network/systems. The first task we have is for someone to review one of our new configurations and see where we can fine-tune it, also want to change how Kamailio communicates to the cus...
I am looking for a freelance IOS developer to create a SIP phone app for my existing business. The app needs to connect to my FusionPBX server, which is already set up. My ideal candidate would have experience in developing IOS SIP phone apps and be familiar with the FusionPBX system. The user interface customization required is moderate. I would like the app to...customization required is moderate. I would like the app to have shared call appearances feature and a clean, user-friendly design. Key requirements: - IOS SIP phone app development experience - Familiarity with FusionPBX system - Moderate level of user interface customization - SHARED CALL APPERANCES - MULTIPLE REGISTRATIONS/EXTENSIONS HANDLED IN THE APP - Clean and user-friendly design. - FREESWITCH AND FUSION PBX E...
This would be to develop and install a small call flow that runs on FreeSWITCH to screen calls. YOU Must know FreeSWITCH and have used it. You will need to install the FreeSWITCH stack and then write custom call flow for it. Integrates with Google Contacts API and Pushover notifications. you should have experience building call flows, databases, webforms/pages etc... this call flow will answer the call if the caller is in the subscribers google contacts # list, you will send the call to a second # if they are NOT in the google contacts # list, you will ask the caller to say their name and company name you will record that and transcribe it you will send the transcription via pushover to the subscriber you will the call the subscriber at a different number and play ...
I am seeking a FreeSWITCH or Asterisk expert to help me set up a new enterprise. The ideal freelancer should have prior experience in this area. The project requires call routing and forwarding features to be implemented. Other functionalities such as IVR and voicemail, as well as conference calling and recording may also be required. Please provide your experience in this field in your application. Automation will be required for this as well. I basically want people to go to my website, sign up, and get started with processing calls. Thank you.
I am looking for a Flutter developer who can help me connect my app with FreeSWITCH via SIP. Specifically, I need the app to have voice calling capabilities, and it should be able to handle 6-10 concurrent calls. Simply, this will be a softphone that connects with FreeSWITCH and it is used for inbound and outbound calls control. Skills and experience needed for this project include: - Proficiency in Flutter development - Knowledge of FreeSWITCH - Experience with voice calling capabilities - Ability to work with existing designs and implement them into the app If you have any suggestions or ideas for the app, I am open to hearing them as well.
I use a fusionpbx softswitch and I want help in overriding or replacing the hangup cause from 480 to 503 and I couldn't do it, pls bid only if you're a fusionpbx or a freeswitch expert
