Sip proxy support tls srtpJobb

I am looking for someone to give proxy interview for Tableau Developer position.

...In general we are interested in the level that implies a lower number of transactions, which in principle is the most basic and is the one that fits with the startup models. TLS 1.2 certificate for communications encryption (required for PCI certification) Components: - Web control panel (custom Backend). Hosted by Amazon AWS - iOS Mobile App (Not yet

...for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a Country Manager that will manage a team of sales representatives, monitors and trains them, and functions as a 2nd level support when necessary. In the

...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server

I have a new digitalocean droplet loaded with CentOS. I need NGINX loaded and configured as a reverse proxy with the correct x-forwarder header so it can be used behind cloudflare. Traffic => Cloudflare => nginx proxy => origin server PROJECT IS MARKED URGENT PLEASE DO NOT BID UNLESS YOU CAN START ASAP. Thank you!

I have an Asterisk server for private use already running, and would like to acid...server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

Hello, i am getting some errors while trying to run my squid proxy server on my ubunto 17.04 server. Can anyone help?

I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

Hi! I am looking someone who have good knowledge of PHP Sockets and TCP Tunneling. I want make socket connection using proxy. My connection is like below [Logga in för att visa URL] If You can do it, Please bid with word called "Tunneling" So I can know that You have understood what I am looking for

I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

Preciso de um freela bem experiente em hospedagem que saiba preparar um VPS junto com outro servidor proxy. Eu tentei o combo mais fácil e conhecido que é o DigitalOcean + ServerPivot + CloudFlare + Zoho Email mas não consegui. Consegui tudo, menos conectar ao database. Além disso, esse combo não é offshore. Eu quero servidores fora dos Estados U...

Necesito integrar el metodo de pagos de redeban multicolor con woocommerce de wordpress. s necesario que al realizar la implementación hagan uso del protocolo TLS 1.2 a nivel de comunicación. Al igual que si se realizará el consumo del servicio a través de SOAPUI se realicen los siguientes pasos: Buscar donde este el SOAPUI. Buscar el archivo SoapUI-5

I need a proxy creation script and manager for a Dedicated Linux Server that has the following: - Uses multiple provided IPs (Either IPv4 or IPv6) to create proxies (i.e. Allot a specific number of Proxies to multiple users using the pool of IPs provided) - Works on Datacenter Dedicated Servers. - Ability to create proxies can be authorized by both

We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

...tools needed to make the box a complete antispam gateway. Installing proper mail server components for multi-domain future use. - Nginx with SSL (let's encrypt) - Postfix with TLS - Redis for caching - Dovecot with Sieve With SSL, nginx, and all the standard tools that we need. Then we should setup a test subdomain so system can fully route and process

1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

...However, here is the problems that I need your help with to get it to resolved. Problem 1: SMTP Banner Check Reverse DNS does not match SMTP Banner SMTP TLS Warning - Does not support TLS. More Info SMTP Transaction Time 15.598 seconds - Not good! on Transaction Time Problem 2: There are alot of warnings in the Exim Configuration Manager

Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

Need help turning 128 subnet into squid proxy. Also I need proxy speed as fast as possible so squid tuning

i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [Logga in för att visa URL] or yate or besip [Logga in för att visa URL]

I'd like to build a proxy server using Microsoft 's Azure. I could build it before by another person's instruction. But there was something change on Azure so I can not do it in the previous way. I can not solve it because I can not contact the person who taught me last time. I have no knowledge at all. I want you teach me how to do it and just

add sip trunk to elastix i have sip trunk from STC

I am looking for a Male Proxy who is an expert in DevOps . Once finalized, I will be sharing the job description along with other required details. Please find below my requirements Gender: Male Technology/ Skills required: DevOps/ AWS, Chef, Puppet, Linux, Heroku, Jenkins, Docker, etc.. US Time Zone: PST Interview Availability: Must be available to

...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...