Asterisk tls srtpJobb

This is a hobby project ESP32 LTE IOT project. I didn't find good documentation on the TLS for the Sim7000g. I figured this is essentially boilerplate for something someone has already written. So could be an easy project and save me a day or so in programming. Looking to use an ESP32 LILYGO TTGO T-SIM7000G for MQTT data exchange. Looking to use AWS and will require TLS secure communication through MQTTs protocol. The design will require OTA update either through HTTPS or SFTP or MQTTs. All network communication shall be secure and over LTE (not WIFI). Looking for good documentation in the code. All source shall be C/C++. Source files should utilize PlatformIO.

HI would like to add an external IP when calls sent from voip switch to asterisk and add and an external ip which would be media ip and sent back the calls to Voip Minutes.

My GSuite email is no longer sending emails out since a few days ago when I got the AutoSSL renewed. When I try to send any outgoing messages, I get an error with the message: Message not delivered ... TLS Negotiation failed, the certificate doesn't match the host. I use SendGrid for my outgoing mail; and Cloudflare for CDN. I need help with configuring my server mail settings so that it recognizes my mail server SSL and emails can start getting sent out successfully again.

My server has been sending emails without any issues all along. I have however started receiving this message when trying to send mail out: Message not delivered There was a problem delivering your message [***]. TLS negotiation failed, the certificate doesn't match the host My server runs on Centos 7 and I use Clouflare too. Thank you

Looking for Brand Research Specialist for projects undertaken by TLS. visit :

Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

I don't have a problem on Internet Explorer but get this error when I access my website with Chrome The connection used to load this site used TLS 1.0 or TLS 1.1, which are deprecated and will be disabled in the future. Once disabled, users will be prevented from loading this site. The server should enable TLS 1.2 or later.

I have a grandstream model UCM 65 10 that reboots and connects the voip trunks very well. however after some minutes, the registration refresh is not working properly. need help: this is the error message in the asterisk logs WARNING[1923] chan_sip.c: Timeout on 896482223-292013357-1450912169 on non-critical invite transaction.

Hi I need an experienced developer with wallboard design , that understands a call center operation and wallboard requirements also reports , logon and logoff features etc. Thasnk

We have Installed the latest version of issabel with asterisk 16. We Also installed Vtiger CRM 7.1.0 and the connector works succesfully. At the moment click'n'call works perfectly and it's possible to pair an extension with a single vtiger profile. We are also able to see a pop up each time we call a number from vtiger interface, This pop-up only says that the extension received the call. We Would like to have the "incoming call" pop-up in order to see immediately who is the caller or in order to open a new ticket directly from the pop up.

Web Hosting with cPanel and WHM security cPanel/WHM and WHMCS - Server installation and setup. Firewall CSF or any SSL / TLS Support (Free SSL, Let's Encrypt Or Any Paid SSL) Email - setup Web Hosting package(Shared, VPS and Dedicated) creation

Buscamos un programador. Para desarrollar, modificar y agregar nuevas funcionalidades de nuestra central PBX Issabel. Actualmente ya tenemos el módulo call center funcionando. Relación a largo plazo, para tener un programador / soporte mensual.

I have odoo v11 community edition. I am looking for a freelancer who can help me to implement VOIP calls form Odoo. I would like to implement function of click to call in contacts.... I dont have any asterix, freePBX or cloud calls system yet, I do have SIP Trunk lines. I have seen many projects that had integrated odoo with FreePBX system, and I would like to do the same. So in short you will be connecting odoo with our freepbx system. the odoo crm, contact, leads, customers and more, should have click to call function, popup for outgoing call. If you know odoo 11 and you already develop/integrate phone system, please let me know. I need this job to be done ASAP. Thanks

I need expert guys who have good knowlege of opensip and Asterisk

We are running a voip server on Kamalio needed expert to setup TLS certificate and connection as stopped working in ios 13

I am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr without rates for now. Platform must be able to handle thousands of concurrent calls and also doing rtp proxy. Platform must have a did management feature and the ability to provisioning and turn up customers immediately, retail with1 number or wholessle with thousands of numbers. Cdr must be very detailed with sip trace available with the click of a mouse. Basically all the features of the latest version of opensips EXPERTS ONLY. If you have a customized wholesale version of a2b I am willing to look at it also. Maybe doing signaling only with a opensips proxy in front of it

I have an SBS2008 server, running Exchange 2007. Previously my SSL certificate was updated annually, then any EXTERNAL PC's running outlook could conenct remotely to exchange. The last time the certificate was installed this stopped working. You may need to activate TLS 1.2 on the server. I think all updates have been applied, but can't be certain. Now My PC's will connect to exchange when INSIDE my server network or when I connect to my network via VPN, but no longer connect to exchange without a VPN running. it appears to be something to do with the security certificate bindings in exchange 2007 on the server. The SSL is a single domain SSL, not a wildcard. If I needed a wildcard, I will get one (if this is causing a problem with ) I just need someone who know...

I have an SBS2008 server, running Exchange 2007. Previously my SSL certificate was updated annually, then any EXTERNAL PC's running outlook could conenct remotely to exchange. The last time the certificate was installed this stopped working. You may need to activate TLS 1.2 on the server. I think all updates have been applied, but can't be certain. Now My PC's will connect to exchange when INSIDE my server network or when I connect to my network via VPN, but no longer connect to exchange without a VPN running. it appears to be something to do with the security certificate bindings in exchange 2007 on the server. The SSL is a single domain SSL, not a wildcard. If I needed a wildcard, I will get one (if this is causing a problem with ) I just need someone who know...

Hello, I'm running VitalPBX. Using PJSIP extensions with SIP trunks. All with TLS signalling and not encrypted RTP. When placing multiple calls at once, voice quality is horrible and choppy. I need someone who can look into Asterisk settings for me and possibly do a network capture and understand what is going on.

i need a sysadmin to Install A2Billing + Asterisk + FreePBX

Hello i want an expert to install FreePBX + Asterisk and A2Billing.

Searching for somebody who has idea about proxy, TLS termination, cryptography and security in Linux.

I want to create a Voip trading platform (like for example: - See Video ) for VoIP sellers and buyers. The system has to work with Kamailio and Asterisk.

Need someone to configure click2dial to make calls. Calls using softphone and freepbx - done. Odoo is connected to freepbx - done. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. Thanks

Needed to interface with the RAMQ needs to know french Web Service Basic Profile 1. Web Services Description Language 1.1 (WSDL) Extensible Markup Language 1.0 (XML) Unicode Transformation Format 8 (UTF-8) Message Transmission Optimization Mechanism (MTOM) Text Message Encoding 0 Simple Object Access Protocol 1.1 (SOAP) Secure Sockets Layer 3.0 (SSL) Transport Layer Security 1.2 (TLS protocol) Hypertext Transfer Protocol – HTTP/1.1

We’re looking for someone who has extensive knowledge of VOIP services in Odoo with OnSIP. Also good knowledge of VOIP Asterisk server and VOIP integration for Odoo. If you're an expert in this job, please apply and others are not acceptable. Please submit your proposal with your best samples. Thanks

Need a individual that can develop me a connection between my asterisk voip server to a crm system i build in house.....

I must have made the follo... Wednesday, Thursday and Friday at 4pm (4pm) Must activate the following commands asterisk -rx "queue add member Local/28282836@phones to 32212 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32211 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32213 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32214 penalty 7" 3. Script Must be every Monday, Tuesday, Wednesday, Thursday and Friday at 07 (7 am) Must activate the following commands asterisk -rx "queue remove member Local/28282836@phones from 32212" asterisk -rx "queue remove member Local/28282836@phones from 32211" asterisk -rx "queu...

hello, i need someone to help me out about this project. i need a net/http wrapper done which supports the following: http 1.1 / http2 proxy support header order applying custom presets to client hello (with refraction-networking/utls & ja3-transport) please only comment if you are really expierenced in golang, net/http and utls

We have open source voip server and api using linphone sdk in apps to run calls, but ios 13 its stopped working due to TLS connection, My current voip engineer isn't available is there someone who can help?

...website has an expired certificate. however, i have renew it already. the new expired day is 2020-12-09 then i check on whynopadlock, it shows “You have an invalid or missing intermediate (bundle) certificate. This may not break your padlock on all browsers, but will on others. Please contact your SSL Vendor for assistance with this error.” and “You currently have TLSv1 enabled. This version of TLS is being phased out. This warning won’t break your padlock, however if you run an eCommerce site, PCI requirements state that TLSv1 must be disabled by June 30, 2018.” The host is google cloud i use wordpress i install it on google cloud ssh in the first time , then it expired 2 days ago. i just follow the YouTube to renew the ssl on ssh . (just copy th...

install openvpn and fix the connection issue in asterisk and fix some issue

my provider's server crashed (raid issues) and i need a find gentleman who can install freebpx / asterisk and secure it on a cloud server having ssh access...

...we would like to implement dialers and outbound stuff but in a more basic way, not yet to implement predictive or so. Also we would like to have a Webrtc phone integrated in the agent panel that they can interact with. So far we are developing the tool using Laravel, Vue.js and MySQL, the application interconnects with FreePBX to get data and eventually the idea is also use the capabilities of Asterisk/FreePBX through AMI and AGI to trigger actions like make, receive and monitor calls. I would like to see the possibility to talk to you and see if you would be interested to help us with your knowledge and experience in the VoIP related area. We are looking to give to our customers a really nice tool to use and we are open to suggestions if that would be the case and make this pro...

Project A back end server to forward client requests/replies to/from mongodb with additional functionality to provide real-time updates to the client. Keywords: Backend, Django, websocket, mongodb, AWS Ubuntu. Setup: Mongodb server (MS) on AWS Ubuntu Django web server (DS) on AWS Ubuntu Multiple mobile and desktop web clients (for testing) TLS/SSL for MS<->DS SSL, WSS for DS<->client Data flow: Client is prompted to enter a client session through login/password . The DS has to authenticate Login/password pair on the MS and make a connection. If the connection to MS is successful the client can setup a websocket connection and send requests over it: RunWatch, StopWatch, Query, Update DS forwards the requests to mongodb and sends back the repl...

I have an asterisk server, the calls are droping as soon as the call reaches the IVR, I require help to fix this problem urgently please.

We have a Asterisk/FreePBX-system with 2000 preconfigured extensions. We need a software to run on another Linux machine that continuously monitors the status of the Asterisk/FreePBX system state and triggers email-alerts if the server is unresponsive. The software also need to log in a local SQL DB when a PJSIP-extension/endpoint on the Aserisk/FreePBX changes state. Details of the Endpoint should be logged with the state change and email alerts should be able to be configured on specific extensions and state changes. (I.e Unavailable->Available/"Not in use" and Available/"Not in use"->Unavailable). The software needs to send heartbeat email alerts that it's running and have access to the Asterisk/FreePBX server. All email alerts need t...

We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single message then the user will be forced hangup.

our Gsm Gateway software use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

Looking for someone who can build a website similar to Patreon or other crowdfunding websites but on a much smaller scale to start basically as simple as possible. Must use latest TLS etc. Must be able to take Paypal, Visa, MC, Discover to start. Thanks. This is very early stages and I would want to go live Q1 2021 if able to.

We need an iOS and Mac OS developer with good knowledge in network protocols (TCP, UDP, SSL, TLS, etc). The network protocols knowledge is a must. Further information will be shared in the chat. Thanks

Hi, We are looking Two Way SSL handshake solution. If you have done before two way SSL/TLS implementation then bid on this project please.

**To do without any commercial module** I have a requirement of Asterisk/ FreePBx to make it redundant. for that what my client is looking for is One main PBX at Location A and another PBX box at Location B. In case Location A goes down for any reason then Location B should kick in and continue the phone service for the company. Is this is something you can do? Please let me know.

Petite PME recherche spécialiste Exchange pour assurer la maintenance de son serveur Exchange (pas réalisée depuis plusieurs années). Voici le détail des missions attendues : - Maintenance des bases de données Exchange afin de garantir leur intégrités et apporter les modifications si nécessaire. - Sécuriser l’envoi des emails via une connexion sécurisé (SSL/TLS) - Faire un point sur l’ensemble des comptes Exchanges existant pour supprimer les comptes obsolètes, désactiver les comptes inutilisés et archiver les emails qui ne vous sont plus nécessaire au quotidien dans vos boites emails - Faire un nettoyage des activations de comptes faites depuis les mobiles (...

For custom made Invoices to bill clients for incoming calls through our Asterisk PBX or for items sold and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change templates to pdf without loosing formulas and calculations (Foxit ?, etc.) 3) Linux 4) MySQL DB 5) PDF 6) PHP 7) KVM qcow2 (if possible, for disk management) We use background images in LO Calc spreadsheet, to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image ! This project is not for beginners !! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This project...

We are looking for a Call Filter solution for Asterisk .Call Filter will Whitelist and Blacklist numbers using rules.

...use their domains. It's like how Tumblr does. These domains may be a wild card, or just a full domain. It's easily possible if you just omit the server name from Nginx config, but when you want to combine it with SSL it's a big headache as SSL should be issued, and applied to the server on the fly (when you connect to the website). 1. I tried to use Caddy () They do have an option that allows you to issue SSL through "Let's encrypt" on the fly, but working with wildcards is a bit complicated. 2. I tried the Lua version of Let's encrypt that puts another layer onto the Nginx, but I prefer to keep using the Nginx framework, and it doesn't seem to work as I expected. 3. I also used ZeroSSL API to issue SSL on the fly using an API

I have an asterisk server which has oem set functionality which I want to be accessible via an android app. Below are the requirements. Allow users to be able to call each other within the netowrk via UserID - users must be able to call each other at anytime i.e. constantly connected to the SIP Include speech synthesis in the app - so user can type in what they want said at the start of the call and it gets sent to server Ability to set Caller ID internationally View call history Set voice pitch dynamically through app if possible Receive DTMF dial tones as sent to server by callee Must be modern design and run smoothly. Must include FAQ with details on how to use the app Must include help page with contact information On startup: must include a progress bar with audio file play...

Create a JavaScript program that will calculate the volume of the cylinder. The formula needed to calculate the cylinder volume is: V equals r squared asterisk times pi asterisk times H The height (H) and the radius (r) are arbitrary, while Pi must be defined as constant (3.14). Record the result using the alert method or document.write. Note: Send the work as an html (zip) file.