Sip tls jobs

...proof of concept (POC) we want to have SSL&TLS port listening for connections and logging transfered data to console and logfile This port terminates the SSL/TLS connection and takes the packages, traces them into log and sends them further as a SSL/TLS client to the IMAP server thunderbird ----> your SSL/TLS socket/endpoint ----> logic with slf4j to trace packages to log ----> open connection to target imap server via SSL/TLS like the thunderbird requested it Your task is to implement the POC, which allows any client to login to your implemented IMAP server. Your server does: - start a local port (local socket) for terminating the thunderbird SSL/TLS connection - allow access from any client to your port - trace the packages to ...

we are an intercom company with hardware that makes video calls to an ios app. The app simply receives the video calls, and it runs on linphone client. For our iOS app, we have some users having an issue loading the video call correctly. it hangs at the loading screen and video does not load. (only some users) We would ...that makes video calls to an ios app. The app simply receives the video calls, and it runs on linphone client. For our iOS app, we have some users having an issue loading the video call correctly. it hangs at the loading screen and video does not load. (only some users) We would like this issue to be fixed. Requires a person with good knowledge and experience with iOS development and linphone SIP client. This opportunity may lead to a permanent position with ou...

Aplicacion que corra en linux y/o Windows, que realice una comunicacion dicom local con los equipos medicos locales. y que reenvie los estudios a una pacs instalado en una vps. principales caracteristicas: Agregar Campo DICOM / Asignación de etiquetas / Morphing (segun el destido remoto) Capacidad de TLS para DICOM seguro a través de Internet Enrutamiento basado en reglas Dicom Query/Retrive Agregar un SR (PDF) antes de retrasmitir las imagenes al destino final con un Formulario que pueda modificarse segun el caso Poder reinstalar e instalar el softwere las veces que sea necesario y equipos necesarios Referecias

Please respond, with a quote, if you know how to fix the following problem: You have an SSL/TLS certificate from AWS Certificate Manager in your AWS account that expires on Jan 05, 2021 at 12:00:00 UTC. This certificate includes the primary domain and a total of 1 domains. AWS account ID: 431240572300 AWS Region name: us-east-1 Certificate identifier: arn:aws:acm:us-east-1:431240572300:certificate/f21aceb8-2834-4c64-a924-c873abf42e77 AWS Certificate Manager (ACM) was unable to renew the certificate automatically using DNS validation. You must take action to ensure that the renewal can be completed before Jan 05, 2021 at 12:00:00 UTC. If the certificate is not renewed and the current certificate expires, your website or application may become unreachable. To

Hello, We're a small company of IP telephony professionals located in Ukraine looking for a good partnership with an EU citizen or company of the same scale. We would like to establish a partnership with you for IP telephony outsourcing, SIP trunking, and DID (virtual) number provisioning. Message me if the above sounds like you. Thank you for your attention, and have a wonderful start of 2021! Dmitry

Need a routing expert to set routes for 3 interfaces. First interface is main network using as gateway () Second interface is VoIP for phone using as gateway () Thirth infterface is direct connection to SIP using as gateway (ip ) and needs to contact

I have a Freepbx system that has been up and running since March of 2020 with no issues. All of sudden our customers can't dial into to the SIP trunk. The ISP is claiming there is a "500" error on the network coming from our Freepbx box. Need help diagnosing this issue.

i want vpn expert developer . expert mobile networking vpn . free connection vpn . expert protocol and tunneling . vpn client login with user and password . contact me only vpn expert developer protocol list 1- hhtp2 2- DNS 3- proxy 4- ssh 5- ssl 5- L2tp 6- ipsec 7- tls 8- webrtc 9- ikev2 all protocol working automatically . with tunneling

I need a sip based webrtc webphone with all basic features and audio video calling plus instant messaging is desirable but not must

Lindoor is linphones/Belledonne's latest open source SIP SDK esspecially developed for use with SIP video doorintercom systems deployed in multiple dwelling units. We want Lindoor to use a html link to load a specific landlords logo, change colorscheme from orange now towards blue and all English text changed toward by us supplied Dutch text. The automated Provisioning assistant wizard when popping up after first start of the APP needs to popup towards the QR scanner functionality. All provisioning settings included html link for logo will be imported from the provisioning link that was embedded in the QR picture (no coding required). The APP will be compiled and delivered to me running in both Ios as well as Android. We will publish the supplied APPs in Appstore and...

We are looking for help integrating a rather simple SOAP api. We are looking for someone that could build a wrapper for transmitting data to this api in JavaScript (NodeJS) if possible. The api is used to send a CSV file and a couple of values to an endpoint. The endpoint requires TLS authentication. We can provide a test certificate and other test data.

We are looking for an experienced artist to design and create artwork for our new Paint and Sip Studio. All paintings will need to be done using Acrylic Paint. All the specified details will be provided. We are after 15-25 different paintings and designs. We have a lot of inspiration and ideas to make this a simple process for you. Videos of you painting will need to also be created as a teaching method for our other artist. (these videos will NOT be used publicly) Thank you,

I am in need of expert to setup SIP for our TWILIO IVR system. I currently have Twilio IVR system that was designed by my previous IVR freelancer but he is unable to setup SIP feature for incoming and outgoing VOIP calls. I need someone who can set this up.

Looking for some assistance setting up a SIP trunk on our Mitel 3300 MXe controller.

Need help how to create a encrypted phone that connects to my sip server

i am looking open vpn developer . need many many protocol . i want unblocking free data vpn for my client All protocol working automitic this vpn for free data using . 1- proxy 2- Dns 3- http2 4- ipsec 5- sstp 6- ikev2 7- icmp 8- webrtc 9- wireguard 10- TLS 11- SSH unblocking imo whatsapp and many many apps I send you example vpn link this all vpn working here free data . This is my current vpn

...screen and therefore we need a speech 2 text functionality in jitsi. In the jitsi/jigasi documentation () is mentioned that it is possible to implement: "It is also possible to use Jigasi as a provider of nearly real-time transcription while a conference is ongoing as well as serving a complete transcription after the conference is over. This can be done by using the SIP dial button and using the URI jitsi_meet_transcribe. Currently Jigasi can send speech-to-text results to the chat of a Jitsi Meet room as either plain text or JSON. If it's send in JSON, Jitsi Meet will provide subtitles in the left corner of the video, while plain text will just be posted in the chat. Jigasi will also provide a link to where the final, complete transcript will be served when it

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I'm looking for a VOIP/FIREWALL FAILOVER expert. We have a call center, approx 200 reps and we dial our customers based in the US via VOIP. We already have 4 different bandwidths deployed and already have a firewall instal...our customers based in the US via VOIP. We already have 4 different bandwidths deployed and already have a firewall installed which is Pfsense. Currently, we dial out from 1 bandwidth and all calls get connected from that single bandwidth, the problem is when that bandwidth goes down or we face a high latency issue, our calls go drop and we have to manually switch our calling traffic which is SIP traffic from port 5060 to another bandwidth. So, we need a solution preferably on Pfsense or any other firewall where our calls do not go drop even if the ban...

Hi, we are looking an developer who can develop an sip/voip softphone for android with custom screens thanks

I host an entertainment show called Trap Poetry Sip n Write where people can enjoy writing poetry while listening to classic poetry through Hip-Hop. I am looking to have the logo by the end of 2020. I have attached the flyer that I use and am looking to take the mouth part and turn that into a logo.

I need someone who can install Vicibox Server on Cloud and setup SIP account on that for inbound and outbound call center. Skills Required Linux Asterisk PBX VoIP Call Center CentOs

I need someone who can install Vicibox Server on Cloud and setup SIP account on that for inbound and outbound call center.

Paint Me Silly Too is a small business being launched in 2021 which sales painting supplies for a group in a box. For local customers there is also a bakery component to sale customized cakes. Slogan: Paint Me Silly Too- Party in a Box I would like the logo to be colorful and symbolize both the art and the cake component possibly popping out of a box or illustrates a su...which sales painting supplies for a group in a box. For local customers there is also a bakery component to sale customized cakes. Slogan: Paint Me Silly Too- Party in a Box I would like the logo to be colorful and symbolize both the art and the cake component possibly popping out of a box or illustrates a surprise factor. I would like these packages used for events such as: family fun; paint and sip; children pa...

We have a Cisco UC 560. We bought a Fanvill i20s Voip Door Phone. We need someone to get it connected with the Cisco UC 560. TeamViewer Available

I can't see this page Enable TLS 1.0, TLS 1.1 and TLS 1.2 in Advanced Settings and try connecting to again. If the error persists, the site may be using an unsupported protocol or cipher suite such as RC4 (link for detailed information), which is not considered secure. Contact the site administrator. I.E. works fine in windows 10 and sites work fine with all other browser.

Hi Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Olá Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hello Freelancers, We want to develop a Windows 10 Based SIP and XMPP client software. Basically we need a similar software like Bria. The basic functionality of the software will VoIP soft phone and XMPP chat functionality. You can use any opensource framework to start with. It should have different type VoIP codec support. We have FreePBX as VoIP server and Openfire as XMPP server. This software will work as client to them. Regards, ALSOFT

I need a webclient class that implements WS-Security signed with wss cert and TLS authorisation using TLS cert. No need for UI just a proper communication with the service with WSDL and XSDs: A partially working SOAPUI (test data out of date) project available. A brief description from documentation: All network services are secured using WS-Security. Client system is mandated to use pair of certificates provided meaning cerificate to use system (TLS) and certificate for authetication of data (WS-Security) To use system you need to : 1. Create cecure TLS connection using TLS certificate 2. Prepare proper XML message according to provided WSDL/XSD 3. Prepare SOAP envelope with WS-Security header (XML signature

...calling use a) Agent browser conencts to freeswitch over WebRTC, but call is sent out on SIP to our SaaS platform for call termination b) Agent browser connects over WebRTC and call is placed directly over SIP from freeswitch to SIP trunk provider. During the active telephone call we will require an API or trigger from Agent side to force the telephone call to become a conference call. Audio between the agent and the trunk will be muted for DTMF tones. A third SIP call leg will be added to the existing call connecting the audio path between the customer and third call leg parties. DTMF and audio will be active between customer and third call leg. Third call leg will be sent a XFER header in the SIP information. XFER header will be pushed with the API...

Hello dear ..we are looking solution for mobile phone make a like voip gsm gateway for android app,, call will come from sip server to android apps .. then apps will sent call to gsm dialing ,,

Freelancer will setup Asterix - SIP server for Simbank and GOIP. Please let me know about your experience first.

I need a Simple SIP Registration test app for Windows. It will mock an Asterisk server receiving REGISTRATION requests from SIP clients (Softphones or Hard Phones such as Cisco, Polycom, etc.) and it should reply as Asterisk does and it should show the entire dialog and optionally log it to a file. A simple GUI is enough, where I can create an extension (number and password) and then configure a phone to point to it and try to register (the App would get the local IP of the machine it is running on - if more than one is detected, I should be able to chose/select which ones to bind to and listen on). The listening port should also be configurable (default to 5060 but changable). The app should remember the last configuration, screen size and position, etc. I need a standalone...

...ENGINEERING Exam. I added a sample questions and subject. I need somaone that can solve this kind of question. The exam is tomorrow. Subject Week Date Subject 1 05-09.10.2020 Intro to microwave engineering 2 12-16.10.2020 Transmission line theory 3 19-23.10.2020 Transmission line theory & Smith Chart 4 26-30.10.2020 Transmission line theory & impedance transformation 5 02-06.11.2020 Types of TLs, Microwave network analysis 6 09-13.11.2020 Microwave network analysis 7 16-20.11.2020 Impedance Matching 8 23-27.11.2020 (Withdrawal Week) Impedance Matching 9 30.11-04.12.2020 Impedance Matching 10 07-11.12.2020 Power dividers and couplers 11 14-18.12.2020 Power dividers and couplers 12 21-25.12.2020 Microwave filters 13 28.12.2020-01.01.2021 Microwave filters 14 04-08.01.20...

for one of our projects, we are having to establish secure TLS connection with a server and come up with a symmetric key after negotiations with the server for further bulk encryption / decryption of session data exchanges. This needs to be done on ESP32. Freelancers with prior experience with such security implementation may apply.

sip voip website to Mobile call

Buscamos un perfil experto en infraestructura de Microsoft Teams, capaz de montar un SIP Trunk para unos 10.000 usuarios.

I need a asterisk help .I want to know the wrong password. if any sip user is trying to register with wrong password on my asterisk server so how i can see the wrong password or what password they are using..

...(not secure ) server. 020-11-21 10:27:35.890716+0200 argirl[553:43948] [tcp] tcp_input [C31.1:3] flags=[R.] seq=1449662732, ack=4131255208, win=119 state=ESTABLISHED rcv_nxt=1449662732, snd_una=4131255208 2020-11-21 10:27:35.896254+0200 argirl[553:43948] Connection 31: received failure notification 2020-11-21 10:27:35.897250+0200 argirl[553:43948] Connection 31: received ECONNRESET with incomplete TLS handshake - generating errSSLClosedNoNotify 2020-11-21 10:27:35.897447+0200 argirl[553:43948] Connection 31: failed to connect 3:-9816, reason -1 2020-11-21 10:27:35.897555+0200 argirl[553:43948] Connection 31: encountered error(3:-9816) YOU CAN SEE ERRORS IN THE END OF LOG. But I think that is nonsense. Bcz its happened on all cloud platform include my own server. You can see full ...

...classes or ask for more information (so 2 buttons) similar to the booking option at the bottom section of this website. I need all the information on the website easily findable to all kinds of customers and I also need the store to look better and make people want to buy the art there. My basic idea is the home page will have all products and services (paint and sip party, art classes, team building, illustrations, murals, blog sneak peek, store, after school classes etc) in the template of the first website with the ability to click for more information. To further explain, each of these options should be clickable and then have option to buy products or enroll and pay for classes like in second website. I also need all of the current information on the website

i have Ubuntu 18 VPS and i want to configure it with Exim4, dovecot mail. Conditions and info: 1- Server already have a WordPress website with Nginx, Mariadb, php which i want to keep. 2- Developer should configure the mail server as below: - All mail boxes should be in /home/mail/$domain_name/$user/mailboxes - POP3/IMAP should have the default ports (SSL/TLS, non-ssl) to configure it smoothly with any mail clients such as MS. Outlook. - mail servers (incoming + outgoing) should be: OR - Install RoundCube on (webmail.domain.x) 3- Developer must provide a complete (clean) shell commands with configurations made to have this job done, splits with function name like: #installing Exim4 $ sudo apt-get install ... $ sudo systemctl ...... $ sudo .......... # configure

...16 idle, 5/0/0/0 working, job queue: 0/0/0/0, scheduled: 7 loaded plugins: charon aes rc2 sha2 sha1 md5 mgf1 random nonce x509 revocation constraints pubkey pkcs1 pkcs7 pkcs8 pkcs12 pgp dnskey sshkey pem openssl fips-prf gmp agent xcbc hmac gcm drbg attr kernel-netlink resolve socket-default connmark farp stroke updown eap-identity eap-aka eap-md5 eap-gtc eap-mschapv2 eap-dynamic eap-radius eap-tls eap-ttls eap-peap eap-tnc xauth-generic xauth-eap xauth-pam tnc-tnccs dhcp lookip error-notify certexpire led addrblock unity counters Listening IP addresses: Connections: test: %any...18.x.x.x IKEv2 test: local: [CN=VentanaTest] uses public key authentication test: cert: "CN=VentanaTest" test: remote: [18.x.x.x] uses public key...

ESP32 Mongoose - OS device design (PCB and soft or soft only). GCP, OTA, MQTT, TLS, UART, AP, ACL, JSON etc. Knowledge of MOS is essential Initiative and ingenuity needed

Hello, Would like to have a good troubleshooting tool for VoIP services. While working with clients on initial provisioning of our service would be able to verify all the network latency between client location to our DC and vice versa. There are several things involved to ens...negative), latency, packet loss, round-trip time (Ping), Mean Opinion Score, and packet sequence errors. These factors are contributions to a Quality of Service (QoS) status check. 6. Analyze the Ports and Network Traffic blocked by Firewalls VoIP. 7. I have some basic tool for an example for you 8. Just to ensure the list Latency Packet loss Traceroute Speed (Download and upload) MOS Jitter Port scan (SIP and RTP) Option to select server to preform tests Recommended Latency, MOS, Jitter Send report an ema...

I am looking for an audio , SIP , WEBRTC, EXPERT to build a product like with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all. I will provide sample layout of homepage and dashboard to NOT CONTACT ME IF YOU DONT HAVE SAMPLE OF SIMILAR PROJECT TO SHOW ME. If you are lazy

I am looking for someone that can install an installation of FreePBX on my Amazon Web Server account. Once installed I need a GOIP hosted on our office network to be linked with the FreePBX to act as the gateway for calls outside of the network and setup 3 SIP lines.

We have a single channel SIP account with Skype Connect (Skype Manager) as a trial, more channels will be added. We want to integrate to our asterisk server to make outgoing calls via Skype. Please, we only interested those freelancer who have done it before and not those wanting to experiment. Project is only consider done when we can make calls from SIP phone to PSTN number via Skype Connect Thanks