I'm in need of an experienced ASTPP/Freeswitch developer to create an Interactive Voice Response (IVR) system for my project. The primary goal of this IVR system is to Play a pre-recorded message when a customer calls a DID, then route the call according to any routing rules configured on the dialplan. This means that the system needs to be designed to handle customer queries and direct calls to the appropriate department. Ideal Skills: - Proficient in ASTPP/Freeswitch - Experience in IVR system development - Strong understanding of customer support processes - Ability to design system for call routing The project is expected to be completed within a reasonable timeframe. A successful candidate will be able to understand the nuanced requirements of the system and d...
When importing customers via CSV file in ASTPP version 6.0, both postpaid and prepaid customers balance always uploads 0. The task is to find the corresponding php file/path on the server and correct it, so that the balance uploads correctly when uploading the customers on the ASTPP both for prepaid and postpaid customers. Positive balance must upload as positive balance. Negative balance must upload correspondingly. 0 uploads 0. You need to make a detailed document how to do that on any ASTPP server, so that someone without technical knowledge takes the document, follows it and gets the same working on the other ASTPP server. Example of the document: Someone from our side will try to upload customers on the provided server to make
When importing customers via CSV file in ASTPP version 6.0, both postpaid and prepaid customers balance always uploads 0. The task is to find the corresponding php file/path on the server and correct it, so that the balance uploads correctly when uploading the customers on the ASTPP both for prepaid and postpaid customers. Positive balance must upload as positive balance. Negative balance must upload correspondingly. 0 uploads 0. You need to make a detailed document how to do that on any ASTPP server, so that someone without technical knowledge takes the document, follows it and gets the same working on the other ASTPP server. Example of the document: Someone from our side will try to upload customers on the provided server to make
looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server 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
We looking for a freelancer who can assist us with configuring STIR/SHAKEN on our existing Freeswitch / ASTPP setup. Here are some details about the project: - The freelancer does not need to have the necessary certificates and keys for STIR/SHAKEN implementation as we will provide the necessary information. - Our Freeswitch / ASTPP configuration is already fully set up, but it needs modifications to support STIR/SHAKEN. - We are open to suggestions and do not have any specific requirements or preferences for the STIR/SHAKEN implementation. Ideal Skills and Experience: - Strong knowledge and experience with Freeswitch configuration. - Familiarity with STIR/SHAKEN implementation and protocols. - Ability to make modifications to the existing Freeswitch setup to support STIR/...
instale astpp basado en astpp, el problema que presento es que luego de haberlo instalado en debian 10 no me permite realizar conexiones sip, cabe destacar que desactive el firewalld y fail2ban para ver si era el problema y aun no me permite realizar conexiones
I am looking for a freelancer who can provide full installation, configuration, and ongoing support for the ASTPP system. The ideal candidate should have experience in ASTPP installation and be able to provide ongoing support. (+ API DEVELOPING PYTHON - telegram bot) Timeline: I need the installation and initial support to be completed within a month. Customizations: I am open to suggestions for any specific customizations or features that can enhance the ASTPP system. Skills and Experience: - Experience in ASTPP installation and configuration - Ability to provide ongoing support for the system - Knowledge of customizations and features that can enhance the ASTPP system. IMPORTANT!!! WE NEED UKRAINIAN LANGUAGE AVAILABLE!
ASTPP Expert Skills and Experience Required: - Advanced ASTPP expertise - Experience in providing training and support for ASTPP - In-depth knowledge of ASTPP With Cluster Configuration Project Description: We are looking for an ASTPP expert who can provide training and support for our team. Our current level of understanding with ASTPP is advanced, so we require an expert who can guide us in the areas of billing. Ideal candidates should have extensive experience in ASTPP and be able to provide comprehensive training and support in this specific area.
...Additionally, we require integration of the apps with our existing ASTPP PBX server, with the following desired features: Support for audio calls with various codecs (g729, ulaw, alaw, Opus, iLBC, GSM) Video calls with video preview, supporting codecs such as H.264, H.265, and VP8 Management of multiple calls Call transfer, pause, and resume functionality Audio conferencing by merging calls into a conference Instant Messaging with end-to-end encryption for both one-to-one and group chat Capability to share pictures and files Address Book integration Call History display Presentation of advanced call statistics Echo cancellation feature Quality of Service optimization Ability to send and receive SMS text messages through the ASTPP PBX server Implementation o...
The freelancer will get access to ASTPP server, where the version 6.0 will be already installed. We can provide also Paypal business account email (but not login to it). Developer must setup Paypal live payments and document it. We tried to set different options under "Live Id*": a) our paypal business email - Live Paypal account b) Client ID from API credentials c) Secret key from API credentials None of options seems to work. We always get same error: “Things don’t appear to be working at the moment. Please try again later.” The project is considered as done and will be paid only when: 1. Developer makes the paypal payments work (a change in ASTPP code/config files could be needed). 2. Developer documents from A-Z all steps to setup live payme...
I am looking for a freelancer who can help me with ASTPP configuration and voip setup. Specifically, I need someone who has experience working with FreeSWITCH, as that is my preferred VoIP platform. The project will involve call routing, so experience with this feature is necessary. I only require assistance with software configuration, and do not need help with hardware setup. Ideal skills for this job include: - Experience with ASTPP configuration and VoIP setup - Proficiency in FreeSWITCH - Strong understanding of call routing - Familiarity with software configuration If you have the necessary skills and experience, please submit a proposal for this project.
I am looking for a freelancer who can help me configure ASTPP for my VoIP setup. I have already selected for the Standard Configuration, and I need help with the VoIP setup and the configuration. This includes guidance on doing my own setup. I would like someone with knowledge of ASTPP, as well as experience in configuring VoIP solutions. The freelancer should also understand industry best practices and customer needs. Please let me know if you are familiar with ASTPP and can provide the guidance and support I need to get my VoIP setup up and running.
Telecom billing and routing software ASTPP installation and setup for community and enterprise license - Full setup with sample customer. -Gateway integration -SIP proxy. -Reseller account. -Payment gateway etc.
We have an issue with Freepbx'es not staying registered on our ASTPP instance. Seems to be an issue from FreePBX side but we can't find how to fix this.
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.
Hi, I am getting the source code for an ongoing project. The ASTPP switch already has API's built for the Linphone. The linphone code is pointing to an old switch. Step one: I just need the linphone to be migrated to a new switch. Step 2: 1. Bundle/Packages and DID -users can choose from —#different bundled minutes(ie. 80 mins to destination X & Y for $5, valid for Z amount of days.), ---#unlimited plans to certain destinations* —#user can buy DID (DID will show as caller ID for that user) 2. Caller ID For regulatory reasons, we can't modify the caller ID to be that of another local provider.. -when users call from the App, it need to show caller ID unavailable (until that person pays for a DID, or a bundle that includes a DID.) -Then when...
I want my ASTPP Server to make outbound and Inbound calls from the Sip trunk routed from its second ethernet interface which it can primarily see when you ping directly. but says Gateway Ping Failed on Fs_cli
i have accounts with twilio and signalwire create me a a sip trunk configured to work on my server only outbound dialing for gateway (termination only) must see calls going out live on my server to get paid this should be easy money for the right person need this for now, contact me for additional details ( must know astpp and goautodial v3 ) *******dont ask for other payment method dont ask to talk on different platform dont ask to avoid freelancer fees!!!!!!!!******
Hey Freelancers i want to setup a free comunity ASTPP and also develop own android Mobile dialer connected with my Server from where dialer will collect whole data about. Routes list and rate lisr + integrate mobile payment gateway or TOPUP facilities from shop or buy. Pin frim retailers shop etc
Hi, I am getting the source code for an ongoing project. The ASTPP switch already has API's built for the Linphone. The linphone code is pointing to an old switch. Step one: I just need the linphone to be migrated to a new switch. Step 2: 1. Bundle/Packages and DID -users can choose from —#different bundled minutes(ie. 80 mins to destination X & Y for $5, valid for Z amount of days.), ---#unlimited plans to certain destinations* —#user can buy DID (DID will show as caller ID for that user) 2. Caller ID For regulatory reasons, we can't modify the caller ID to be that of another local provider.. -when users call from the App, it need to show caller ID unavailable (until that person pays for a DID, or a bundle that includes a DID.) -Then when...
...new deployment using ASTPP for billing to be integrated with FusionPBX utilizing the multi tenant IPPBX typically ASTPP will handle the billing aspects for voip termination and DIDs origination (we currently use IDT Express and ) then map it conveniently to the PBX tenant I will surely avail both cloud VMs with the base OS as per you recommendations most likely will be Debian as per the documentation or of you want me to setup boh ASTPP and FusionPBX beforehand that would be ok too and then you will just help me onboarding with the initial setup, configuration, interconnection and integration between both VMs and between ASTPP with both and IDT to be acquainted with the full picture and end to end cycle. please note that I already have ASTPP serve...
I have an installation of ASTPP.. It does not have the Get Add-on icon.. I need to get the Get Addon icon back. I need to verify what edition on the software that I have..
General. Integration of ASPP with Wordpress as web front (plug-in or other suggestions) and Linphone (or other softphone) as Mobile App & dialer. The integration with Wordpress should allow for full use of other Wordpress features (Of importance is a shared or common login) and third party plugins like payment gateways, marketing, forums e.t.c. Full Linphone customization. Features for both App and Web Sign-up and verification email/SMS Destination rate check Show Balance DID purchase Credit Top Up with Wordpress Gateways Push Notifications Call & Billing history
requiero implementar toda la seguridad requerida, en fail2ban y iptables para mi servidor de astpp, le deje la imagen de los log que presento en mi serivdor, requiero alguien que hable español
he instalado astpp, y realizado todas las configuraciones, pero tengo un herro al realizar llamadas desde el, el error que presenta es este : 2022-07-22 23:04:11.925640 96.80% [ERR] sofia_reg.c:2677 Gaterway_mastersking Failed Registration with status Service Unavailable [503]. failure #5. y las credenciales de gaterway estan correcta
quiero implementar la seguridad requeridad ara mi servidor de freeswch echo con astpp. las seguridad requeridas son de escanio de sip, denegacion de servicio.
The istallation of the softswitch will be used to set up a VoIP platform for a voip retail business start. The softswitch will be installed on a cloud server, and will be configured ready to start using and intergration with an app. The last step will be testing it with an app.
you have to install freeswitch with astpp billing panel. from Astpp all uer & DID, trunk will manage. Voip call should work between PC to PC & PC to PSTN.
I have instslled ASTPP which comes with freeswitch I need someone to configure WebRTC clients to connect web phones For example And provide " how to " guide
Dear all, I am seeking ASTPP expert to help configuiring YeaStar PRI Gateway TE100 with ASTPP VPS where he can demonstrate successfull interconnection and billing for both outbound and inbound calls and DIDs
...ayarlamak, bağlanacak cihazlarının konfigürasyonlarının yapılmasını ve kurulmasını sağlamak. - Yeni Müşterilerin isteğine göre, proje adımlarını gerçekleştirmek, gerekli olan dokümantasyonları ve çizimleri oluşturmak. - Müşterilerin istediği değişiklere göre dökümantasyonları ve çizimleri düzenlemek. - ASTPP astppbilling centrali üzerinden fiyat güncellemeleri telefon numara alımı satımı ayarları, fiyat ayarları yapılması. - Yurt içi ve yurt dışı toptan ses hizmeti veren operatörlerle anlaşmaları ASTPP içinde oluşturmak, geliştirmek ve devam ettirmek. - Ses teknolojileri alanındaki yenilikleri takip etmek, çözüm ve hizmetleri iyileştirmek/geliştirmek, ekip i&cc...
We need a person who works on freeswitch lua and asterisks who can work with us to maintain astpp server
I would like 2 functionalities on the linphone 1) New user account creation via linphone registration.... Can it be made to save customer information to a signup database (We would like the new user to be able to create an account from the app to our ASTPP server or a intermediate API database) 2) User be able to enter username and password for the SIP that will register to our sip server(user should not have to enter server details) Looks like most of the work will be on the 1) item........
Good day Looking for a person who can integrate linphone apps registration with ASTPP open source
Hi There we're looking for a 24/7 NOC service provider especially for voice solutions. Like freepbx and ASTPP , please only respond if your company size is > 10 people. Responsibilities: a) Trace routes / 24/7 availabillity b) Communication with providers c) First line support for end users regarding voip issues
Hello I have a astpp open source VoIP server. ASTPP using LUA script as a dialplan. I need to fix one bug in that script. So looking for someone who is very expert in LUA. If you even worked with ASTPP lua scripts then it will more better then you will understand what I want to fix. Note to Freelancer.com: I have post yesterday the similar project. Just can't understand why it was rejected. Astpp is open source and anyone can modify by their own. Please see Please approve the post and you can delete those admin notes
I installed the Astpp Billing but the issue is the inbound calls are failing giving me an error WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'default' for
Hi Nsikak A., I saw you already have a click2call developed for a client, my platform is ASTPP 5.0 with freeswitch. I need the customer to have the option to choose to receive the call on the extension or pstn, and after answering the freeswitch sends the call to the site visitor. Whenever both incoming and outgoing calls are PSTN, they are billed in astpp. It can be in the two scenarios you already have: 1. Click to dial by users 2. Click to call Thanks
intento cambiar la credencial de Event Socket para asi crer un Freeswitch seguro en astpp, pero al parecer no encuentra la direccion de freeswitch, e usado la documentacion de astpp pero aun asi no logro dar con esto esta es la url vim /usr/local/freeswitch/conf/autoload_configs/
aplicar seguridad a el reseller astpp ya que me estan cortando las llamadas cada tres minutos, y encontraron una vulnerabilidad y me estan drenando el balance
Hello, we are looking for a developer that can handle a couple of projects involving Astpp, FusionPBX, and Freeswitch. If you have experience with Freeswitch and Lua, you should be able to handle the projects. More will be discussed once I get an idea of your experience and hourly rate
buenas, me gustaria un teninin completo de la plataforma astpp, la tengo ya instalada pero al configurarla me da error
buenas, e instalado astpp y le he echo todas las configuraciones, ahora tengo un problema y es cuando los clientes me mandan peticiones me cuelga el servidor, dejare la imagen anexa para que la vean
we have our astpp platform and need configure
I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.
I have an old astpp customizations and I need to update my customizations on the new server. I have the file with all the customizations and file edited, just need someone to re-apply on the new server.
goautodial using ASTPP trunk will having problem for autodial. customer answered the call but get silence and it wont pass to agent softphone. within 5 second will drop the call. tested using manual call with no problem with RTP voice complete. i has been try to use other supplier trunk and getting not issue with manual and auto dial from campaign.
Install 2 ASTPP servers following this doc: ASTPP1 will handle billing, and ASTPP2 here will have a role of freeswitch only (handling calls). ASTPP2 needs to be connected to ASTPP1 DB, so that in ASTPP1 GUI we see also ASTPP2. Users from ASTPP1, must be able to call to user connected on ASTPP2. In ASTPP1 GUI, we must see the customers registered on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only); The done job must
