Sip tls srtp windowspekerjaan

I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

I’m looking for an expert of VOIP, SIP, Hack for educational purpose

I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

...Device to Access-point by using PEAP-EAP-TLS authentication. This authentication method is available in windows but not in Ubuntu/Redhat as of now. But AFAIK, Cisco confirms that Linux can connect by using mentioned method. Currently, following is the key components in picture Wired 802.1x with PEAP-EAP-TLS Authentication method. 1) Supplicant ( Embedded

...(GDPR & ePR compliance functionality) WP Rocket Installation and configuration (caching Functionality) WP Security Installation and configuration (Security Functionality) SSL/TLS certificate installation and configuration...

Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

...(by external means) that the ACME server challenge 'http-01' will succeed [log masuk untuk melihat URL][tls-sni-01].loopback should be defined as function (opts, domain, token, keyAuthorization, cb) { ... } and should prove (by external means) that the ACME server challenge 'tls-sni-01' will succeed Error 2: Error: listen EACCES 0.0.0.0:80 at [log masuk un...

I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

I require a new firmware for my GPS+ sensor device. The device uses a sim card and sends data via TCP/IP. The requirements are for a high-security firmware (TLS 1.2, secure boot, etc). the microcontroller used is stm32f103c8t6. More details about the requirements and hardware can be shared.

...even runs successfully, but the request to Google's servers ultimately times out. We need to find out why this is failing. Possible reasons for the failure could include: SSL/TLS connection problems Android environment variables A bug in the gRPC source code So it is possible that there is a problem with the gRPC source code itself, and meaningful debugging

Cisco voice gateway connection to SIP server and PSTN on E1 PRI

I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

1) Move some WordPress projects from Ubuntu 16.04 LTS to 18.04 LTS + l...Win/Android/IOS-Clients (Certificates/Keys installed on AsusWRT-Merlin 384.7_2) 3) Create full documentation, how to replicate installation of above mentioned (esp. if new OpenSSL with TLS 1.3 is released). - Please provide recommendations projects for OpenVPN on freelancer.com

WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

I need the TLS support of my SMTP server fixed. I use PowerMTA as my mail transfer agent and I manage the servers activities with mumara. I tried adding <domain *> use-starttls yes require-starttls no </domain> but it doesnt work, I need help.

please setup tls ftp (secure ftp) as i dont want to give people root lol this is for the itinvestigator project

VOIP ASSISTANCE SIP DEVELOPPEMENT

I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

...quote by remote access for the below requirement MY REQUIREMENTS 1) I have to shift my existing asterisk vicidial server files ( my dialplans, modules , database ,IVR , [log masuk untuk melihat URL] , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to new & better hardware /server Xeon processor 3.2 GHz with latest Asterisk & Vicidial software

1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

Hello, When there are more than 250 users online on my server, the "TLS handshake connection" takes up to 30 seconds to connect making my sites very slow. This only happens with https websites, the http sites load very fast. See live example here: [log masuk untuk melihat URL] ( 12 seconds to load ) [log masuk untuk melihat URL] ( 288 ms to

...server. I have got a problem with connecting to smtp server as I am getting an error "authentication failure [SMTP: STARTTLS failed (code: 220, response: 2.0.0 Ready to start TLS)]" when I am trying to connect from php. I want you to have a look on my server to see what the trouble is and why I cant connect to the server. You must use teamviewer to control

...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [log masuk untuk melihat URL] secret GZD397XU Host [log masuk untuk melihat URL] port 5060 Register String 390415541985@[log masuk untuk melihat URL]:GZD397XU:390415541985@[log masuk untuk melihat URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [log...

This project is only for those freelancer that has existing similar app to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications

...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

Detailed Specifications will be sent once the job is awarded. This will be a network Print Server that will run on Windows Server 2016/2019 x64 and be a shared printer to network clients running Windows 10 Pro x64 version 1809. The client computer will print a letter sized document to the network share printer (running on the Server) which will save

Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...

as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

I need script or program to make book appointment automatically at website TLS contact

I need some changes to an existing website. I need you to build it.

I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [log masuk untuk melihat URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list

Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default

...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-

...provisioning by email address login. Users should instead enter their mobile phone number to register. After receiving an SMS with a verification code the softphone should download the SIP provisioning data from the server. - Create IOS version and deploy to appstore - Create Android version and deploy to playstore Server side works - Install vialer-middleware