Freeswitch dialplan punët

We have a FreeSWITCH load issue with media (RTP), and need help resolving it. We are using FreeSWITCH's latest version. Without Media (RTP): 1500 CC (5% CPU Usage) With Media (RTP): 400 CC (150% CPU Usage) We want to achieve 1000 CC with Media (RTP), and it should not take more than 5% CPU. PLEASE BID IF YOU HAVE WORKED ON SUCH ISSUE IN PAST.

I have project c++ using websocketpp library - this a mod of freeswitch - using websocketpp to connect remote client and send data I need guys expert c++ to fix some typo and suggest me clean code of this mini project because im not family with c++ typo. You guy will checkout code & remote to my computer for fixing & running project

Looking for a consultant to work on FusionPBX (FreeSwitch) with Start Trinity SIP Tester so that we can simulate calls (Audio/Video/Callforwarding) scenarios. FusionPBX is the wrapper around Freeswitch with GUI. Anyone with similar expereince and in Telecom will be preferrable. It is very urgent for a project of 5-6 days. Interested Candidate, please reply soon.

As a developer, I am looking for a skilled professional capable of providing development services focusing on FreeSwitch configuration. No existing code will be provided, so the individual chosen should be proficient in working with the platform without any starting points. I am counting on the ideal candidate to bring knowledge and skill in the development space to make this project a success. I need candidate to develop WEBRTC Server with a webrtc client to integrate with different CX platform. Post completion of the project, I will need proper documentation in order to install WebRTC server internally for my project.

Hi, I have goautodial 4 setup on server with ssl/tls everything is working great except for the carrier dial plan that does not make calls out. All i need is assistance with the dialplan to make sure that the calls can be made from the webrtc phone built into goautodial 4. The software is setup already the solutions looks like it works fine. I assume the problem is with the dial plan. I need someone that can go through the system to make sure calls can be made out using the web phone built in.

Hello, i am looking for someone could help me install call center script. Will be provide full installation tutorial. Installation come with auto file. (One click installed .sh file) Please only take this jobs if you could install immediately. Server is ready. I attached the installation step and file on below...please review

Finishe FreeSWITCH with DialogFlow integration The current status is: - FreeSWITCH installed and configured 100% working - UniMRCP module purchased, installed and working for Google DialogFlow, STT and TTS integration licenses. () The job is: - Finishe the configuration to complete the integration with DialogFlow. It's partially working. Connects, recognizes what is said but does not vocalize the response. Deadline for delivery: Mar/08/2023

Need a custom CDR and DID report page. Please DM me for more info. Please note that a video interview is required before this project begins.

Required administration training on Freeswitch

- Develop CTI pop-up in salesforce and upload it on salesforce marketplace - Use or any other method to manage SIP registration, Call, Recording and other necessary features(Outbound call, Inbound Call, Blind Transfer, Attended Transfer, Mute, Unmute). - VoIP PBX is based on FreeSWITCH Language -Javascript, Node.js, APAX (Or any other needed for salesforce CTI pop-up)

We need an Asterisk 11 dialplan context that: 1. Executes when dialing any number from the extension 1005 and also from extensions 1070 through 1078 and 1081 through 1089 2. Blocks the call and plays the "number-not-in-service" playback file if the number dialed by the extensions above is equal to the result of the query "SELECT mobile FROM vtiger_contactdetails WHERE firstname REGEXP '^[0-9]+$' and mobile = ${EXTEN: -8}". The query is done to a localhost MySQL database

We are looking for freelance expert who can work an a specific set of agenda to explain and deliver it's outcomes

Hi, We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client. Our flow of calls is like this: WebRTC client -> OpenSIPS -> FreeSWITCH The system is deployed on Azure. We are looking for experienced person who has done such work and quickly help us.

What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time (9a.m. ...

Can not send “@” in sms by Dongle Usb Example: I send: test @ I get: test ¡ What im missing in dialplan ? [textmessage] exten => 111,1,NoOp(SMS receiving dialplan invoked) exten => 111,n,NoOp(To ${MESSAGE(to)}) exten => 111,n,NoOp(From ${MESSAGE(from)}) exten => 111,n,NoOp(Body ${MESSAGE(body)}) exten => 111,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) exten => 111,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) exten => 111,n,NoOp(Send status is ${MESSAGE_SEND_STATUS}) exten => 111,n,GotoIf($[“${MESSAGE_SEND_STATUS}” != “SUCCESS”]? sendfailedmsg) exten => 111,n,Hangup()

Freeswitch amoCRM intgration to enable voice call on the amoCRM

We require someone who can integrate php agi in asterisk box to replicate the similar commands to control dialplan as given under below link here, To enable our customers to run their own IVR using API and SDK Architecture using Cent OS, LAMP framework, Asterisk and PHP-AGI User Accounts connects using API and run their own IVR business logic Our VoIP Arch -> Connected to multiple sip endpoints for each users on their account. I need some one who can help me setup an asterisk on linux machine probably in some better datacenter such as aws or azure that could run behind an proxy public ip for client server connection. And than use php agi to develop an sdk that could send and receive rest api and xml commands for controlling ivr dialplans

Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.

We would like to setup Signalwire video conferencing and integrate it with our HoduPBX freeswitch multi tenant platform

MIKOPBX Call routing Training We are seeking a Russian or Ukrainian freelancer with specific experience in configuring and working with MikoPBX. The freelancer will provide training. Should have previous experience in Asterisk Dialplan and configure gsm gateway for incoming and outgoing calls to setup for call center this is open source PBX need to work on it

Traing for Dialplan and Call Routing for MIKO PBX

...a step-by-step guide for configuring FusionPBX/FreeSwitch to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown which will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN ()

We need a so-called "banner information" for our own web-based control panel, which is connected to the FusionPBX / Freeswitch telephone system. This means that you can double click on the visually displayed subscribers and then add text there, which is then visible in the banner. In addition, background color and font color should be customizable. Also a link is to be opened by means of right-click and "open link" or so. The text, which one can write into the banner remains to be seen so long on the surface, until the call is separated. These data like URL, color for the background, font color, etc. should be extractable from the description field from the destination management at Fusionpanel. That means that we enter there a hexadecimal code color for the bac...

I am looking for a lua developer who can help me to customise something in my VoIP server. It's lua script which do the functions.. and also use mysql database. Experience in the freeswitch server will be an advantage. Thanks

We are looking for an engineer who can assist with FREEswitch Fusion PBX. We need to update voice messages for 5x PBX users (Christmas greeting messages). Change extension labels for 5 customers and install new extensions and hardware for 2 PBX users. If you can offer other services around FREEswitch Fusion, we would be interested to learn your skills and what you can offer to us.

IT Service im Raum Hessen/Bergstrasse Virtualisierung auf Basis von Proxmox un VMWare,Firewall Konfiguration mit OPN Sense,Windows Server Umgebung, Docker und Kubernetes, VoIP mit Asterisk und Freeswitch

Hello, we are a hosted voip service provider and until now we have been using FreePBX and Asterisk, we are looking at moving away from FreePBX. We are open to either Asterisk or FreeSwitch, whichever meets our requirements listed below: Good day, We are a VoIP Solutions Provider that is currently looking to move away from our FreePBX systems we host for clients to a Class 5 Softswitch/PBX. I have compiled a list of features we want but don’t currently have with FreePBX and then the top features of FreePBX that are an absolute must have in this development. This would need to be a linux based platform capable on running on multiple dedicated servers with iSCSI storage. Ideally, we would like the platform built on a AlmaLinux or Rocky Linux. We would want the ability to ha...

Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.

WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk

Hi, We need someone who can upgrade our FreeSWITCH and OpenSIPs to the newest stable versions on Amazon AWS. Currently we use FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is 1.10.8 We also need OpenSIPs upgraded to the newest version 3.3.2 we currently are on: 3.0.2 (x86_64/linux) This is a live production server so it will need to be done pretty quick in a couple hours or so. If we work well together I will have many more ongoing tasks involving FreeSWITCH, OpenSIPs, our PBX and other issues, our main telecom engineer/developer was in Ukraine and we have not heard back form him in months. Thank you! Thank you!

Hi Arshad N., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details

Hi Aqs Y., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details

...using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. Our Soft-phones are made with React.js I need a person who knows what they are doing, we also use OpenSIPs so the codecs in OpenSIPs might not be correct but this is just a guess. Can someone solve this for me, I have a hard time getting honest developers here, it seems like everyone says they can fix it, then I waste a week with them and have to cancel and look for a new developers, please only bid if you are truly an ...

Various tasks in freeswitch. Requirements - understanding, communication, desire to progress on the subject. long-term cooperation

i would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in to it use a SIP software in order to answer customer calls. What I want is a convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field and...convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field and last three data pulled out to select to go into the dispatch panel Deploy PBX Create few extensions Create script for take CID of inbound calls check last 3 records, sent CID and last 3 records via api to dispatch app Help to setup tranks and di...

We want to provide our cloud switchboard solution via freeswitch. For this, I would like to discuss the project with people who have experience in Freeswitch and API development.

Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.

i would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in to it use a SIP software in order to answer customer calls. What I want is a convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text fiel...convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field amnd last three data pulled out to select to go into the dispatch panel Deploy FreePBX Create few extensions Create script for take CID of inbound calls check last 3 records, sent CID and last 3 records via api to dispatch app Help to setup tranks and di...

Connect the Web Application with API and Dynamic Data with the Freeswitch / FusionPBX System.

Hi I need a hand with freeswitch: I have a client that needs to send calls to my freeswitch but his switch doesn't have username+password authentication. He is asking me to have my freeswitch accept the calls based on his public ip address. Let me know if you are able to help but only bid if you have extensive experience with freeswitch as this is a security concern. Max 100 euros. Thank you.

...Description • Candidate should be familiar and comfortable with Freeswitch. • SIP Development experience. • Must be aware of Sip and webrtc integration. • VOIP software development. • Good Knowledge in PBX, SIP, RTP protocols. • Worked on Queue, IVR and Voicemail related applications. • Expert in Freeswitch installation, configuration and... • Competent enough to setup daily call limit and concurrent calls Requirements · Software Development experience in Freeswitch, FusionPBX, Opensips, SIP, VOIP, SDP, TDM, IMS, PSTN, Python, Perl, Linux, and Open Source Technologies. · Strong Technical, Logical and Debugging skills with innovative and result-oriented approach ·working experience in Python, Shell, Pe...

Hello Amrit, We are an Italian internet provider. At the moment we are using a custom Freeswitch cloud pbx calle Hodusoft. We need to customize Linphone to work with. Registration is OK but blf doesn't work and we would some feature like an easy provision and contact. Can we discuss about it? M

Freeswitch / opensips / pbx development work

We are looking for someone who can fix certain compatibility issues with of Flutter BigBlueButton app with latest BBB server. In the long run we'll need to include additional features as well. Flutter Code: Make sure you know what FreeSwitch, WebRTC means. Without depth knowledge on them you'd get lost

We have a requirement of Limit management in freeswitch Limit for concurrent call and Max call in a day for per destination

Ju lutemi Regjistrohuni ose Identifikohuni për të parë të dhënat.

mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

Looking for a Class5 Soft Switch based on FreeSwitch with Android and IOS APP. *Account Management *Calling Cards *Rate Groups / Tariffs *Call Routing Strategies *Call Rates *DID Management *Product Management *Invoicing and Billing *Reports and Alerts *System Settings *Payment Gateways

Hi Eremin P., I noticed your profile and would like to offer you my VOIP project. Cloud PBX based on Opensips and FreeSWITCH. This would be a long term project. We can discuss any details over chat.

Our current situation We are a call centre and have been on the market for many years. We operate a classic telephone service / telephone secretariat. We are currently work...whole thing. The briefing also includes ideas on how to implement the whole thing or which existing functions could possibly be used. The operator panel should be available as a stand-alone web application and work together with the telephone system / FusionPBX / Freeswitch. The WebApp will then certainly be able to be connected to the telephone system somehow in a config file. That would at least be good. The script language on which the WebApp / operator panel is based or developed is up to you. We can install Freeswitch and FusionPBX on a Server and set up a SIP Trunk. We will give you ...