Sip softphone tls srtpроботи

Groovy script (Jenkins DSL) development. Kubernetes Application Deployment resource types (deployments, pods, namespaces, ingress, storage classes, etc). HashiCorp Vault and secrets management best practices. JFrog Artifactory and the various artifactory repo types: Docker, npm, pypi, maven, nuget ... storage classes, etc). HashiCorp Vault and secrets management best practices. JFrog Artifactory and the various artifactory repo types: Docker, npm, pypi, maven, nuget The primary dev frameworks: flask, .NET core, spring boot, and express and their associated build processes Azure databases and other data platform PaaS services and light DBA type work. Network firewall troubleshooting. DNS and TLS certificate management. General familiarity with PwC Labs service offerings. Fluent spoke...

Describe on few topics and a few analytical topics. 1) Diffie-Hellman protocol 2) TLS/SSL related vulnerability 3) Anonymity & unlinkability 4) Secure key, Public key, symmetric key

Hello, I have Dolibarr 13 installed, in a VPS that I manage. I want is that when I click on a customer's phone in Dolibarr, I execute the call through a softphone that is installed on the same computers where they work with Dolibarr. It's just that

I want to get a cloud based application developed for managing incoming and outgoing calls using SIP telephony. Also need functionality to initiate conference calling.

website runs on old VPS with Debian 6.0.6 squeeze + OpenSSL 0.9.8o + nginx/1.2.6 + opensource Magento 1.7.0.2, need update OpenSSL to version 1.0.1 at least, for using newest TLS (because clients browser warns that the site is unsecure), there can be some more problems like old Perl, NB! > there is no access to VPS control panel for emergence restore/backup, only root access to Debian

if you can connect vici to bitrix for my client , give me price and time . thnx

OpenVPN traffic over TCP port 443 uses SSL, but the traffic/"connection negotiation" doesn't look like normal HTTPS traffic over TLS and it can be detected and blocked easily by any advanced DPI. Hence, SSL tunneling or obfsproxy or SSH tunnel is mandatory for the connection to be established over TCP port 443 I want an OpenVPN server over TCP port 443. However, This OpenVPN server has to support having the traffic/"connection establishment" to be over SSL TUNNEL or obfsproxy or SSH tunnel For more details, please check this link All the below is required; - configure the server to optimize the speed, since using TCP 443 slows down the speed and the server needs to be tweaked to optimize this. - I should

Running on ESP8266. AP Web Configuration Server that allow to configure: WIFI SSID, Password, MQTT Server Port, Client ID, Broker, Port, Topic, Username, Password CheckBox to choose: Password Enable, MQTT Enable, TLS Enable For Rx and Tx we will provide the code and its not configurable. Sketch to compile and upload by using Arduino IDE.

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wakeup and receive call. We can make this flow work with some device but not all device work smooth, some device can not wakeup ( android, ios too) i think the app like that ( voip ) need some special skill and tech to implement. My requirement: please see the attact file

i want someone to configure 20 Channel SIP Trunk in my 3cx Server

web base sip dialer i need to develop

I have a few extensions behind a pfsense firewall that can’t connect to my hosted FreePBX. I need help configuring the firewall for this to work.

Looking for Engineer to help configure SIP trunking to ITSP and CUBE

Mobile softphone dialer for android and IOS. Voip enabled using our sip server. You will need our SDK for android and Java to develop it fully. it simple and straight forward.

Configure SSL/TLS on Amazon Linux 2

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I have FusionPBX which works on Freeswitch. I have discovered a bug in the system. 1) When normal external call is made from an extension number in fusionpbx, the sip header contains the extension number and DID number. This is fine. 2) When call forwarding is activated the extension number and DID number are no longer in the sip header. This means that the call cannot be authenticated at extension level when sent to the carrier. I need this to be fixed so that the as a bear minimum the extension number is present in the sip header when call forwarding is activated. Example is attached

...different sizes of mobile devices etc. 12 Google Tag Manager & Google Analytics to be installed, so management can have access towards the websites data and use that data for upcoming marketing events, promotions, blitz sales etc. This data can also be viewed from the chosen marketing company for future marketing needs. 13. After the website is completed a speed test will be conducted. 14. SSL, TLS & HTTPS will be included for the websites security and encryption. 15. A video showing how to manage the back end system of the website. This video is for management incase they want to make changes themselves or have a cost effective solution for the management of their website. 15. A digital certificate will also be included to solidify the websites security and encryption ne...

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We need a softphone that will work with our pbx systems. It needs to work on 3 platforms; IOS, Android, Windows. You can see the detailed information below; • “Sip Server – Username / Password” based login screen - (We require this for the current project. After this project we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must a...

... Ec/Io, RSCP, RSSI, PSC, RRC State,TX Level, Neighbor Cell Measurements, HSPA+ DC in use, Audio Codec Type, AMR Bitrate - 2G/GSM: Band, RxLev, RxQual, C/I, TxPower, ARFCN, RRC State,Serving Cell measurements, Neighbor Cell Measurements, Audio Codec Type, AMR Bitrate - 4G/LTE: Band, RSRP, SINR, RSRQ, RSSI, EARFCN, RRC State, - 5G NR: Band, SS-RSRP, SS-RSRQ, SS-SINR, BeamIndex, SS-PCI - VoLTE/RTP/SIP Layer - Lock NV Item Chipset based api - Layer-3 Messages - GSM RR, WCDMA RRC, CC, MM must be captured and written to a plain text file. The programm must run on Qualcomm based phones. Requirements for participating: - Understanding of GSM, UMTS, LTE & 5G NR technologies - Understanding and experience of Layer 3 tracing and decoding - Experience with Qualcomm chipsets - Providing a...

I have a 20channel SIP TRUNK from Tata.. I want to configuration it in my 3cx system.... and aalso enable API of 3cx for my crm

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I need to install a stable version of Opensips 3 with a web interface on a server running Centos 7. It will be used as an SBC to hide the network topology, change sip headers and load balancing.

Need an expert to install a phone extension system. I have an office but only one receptionist works there with one phone line. I have more business and I would like to use each business with one individual extension number like e.g. accounting 1234512345 / 1001 ext. and monitoring 1234512345 /1002 ext. If somebody call us, the system picks up the phone and request extention number. Based on given ext. number by caller the phone display shows the pre-programmed business name to that ext. number and receptionist can see what business is called (e.g. "accounting" call). Task: advise what system I have to buy and how I have to program it. Thank you

I have an old android app that needs to be updated, however, I can't figure out how to replicate the HTTP/TLS to pull the correct data from an API So, I need someone who's an expert in the ways of packet monitoring, to help me replicate this request I would love to replicate it in Python 3+ if possible

I need to setup call center software and multi tenant fusion pbx as sip server with soft huawei HG8245 as client gateway

2 offices located 1 hour away from each other. Raspberry Pi4 with 3cx PBX at “main” location “Main” location has fiber optic internet 200/60 down/up speed, other location cable 150/50 down/up Public IP but not static Each location has Gateway VOIP Grandstream 4 Fxo GXW-4104 Each location has 2 landlines and 1 SIP/VOIP line Various phones to connect, most are Grandstream GXV3240, 3370 and 3275 No operator per se, small business where whoever is available takes calls Both locations have battery backup on equipment I want to connect all lines from both locations so people at 2 offices can call out, answer, intercom between phones. Ability to take a phone home and be connected to office for the casual employees or for covid lockdown. Make sure phones are pro...

I have an old android app that needs to be updated, however, I can't figure out how to replicate the HTTP/TLS to pull the correct data from an API So, I need someone who's an expert in the ways of packet monitoring, to help me replicate this request I would love to replicate it in Python 3+ if possible

We need a ringless voicemail system created for us. What's the best deal you can give us and what are the actual sending rates for it? Does SIP help to cut down the costs compared to using something like Telynx?

Set up CISCO UC set up in the developer sandbox at Hook it up to out SIP line via twillio and set up VXML and MRCP to communicate with a dialog engine. We will supply the ASR and TTS and the NLP end points.

Hello I would like to know if you can do the following, I have Jitsi Meet 2.0 currently in a vps in ovh working well and I just installed issabel 4 in another vps both are seen by local ip, I already have the annex, sip trunk, for integration with jigasi, and enable the audio transcription to be displayed as subtitles, I also have the dial plan, you can make it work, thanks

Hello I would like to know if you can do the following, I have Jitsi Meet 2.0 currently in a vps in ovh working well and I just installed issabel 4 in another vps both are seen by local ip, I already have the annex, sip trunk, for integration with jigasi, also the dial plan, you can make it work, thanks.

I want to set-up internal communications through a PABX or a Dialer that supports SIP lines. Need the following features: Leads Management Call transfer Mobile Calling facility Live dashboard Call recordings Reports and analytics Real-time notifications Missed call services Dynamic call flows Please let me know if someone can help me out with this. The person needs to be located in Mumbai. Thanks.

App Like Digital Showroom. (Relipay, , Fintech, AEPS, DMT, Travel, Tourism, Recharge, BBPS, Pan card, Bank account opening, G2C Services, Insurance, Loans, Credit card, SIP) Location: Delhi, Noida, and Ghaziabad

I need a Panasonic NS700 telephone system factory resetting and then reprogramming with 6 SIP trunks for 6 total users. The system is located in Nottingham.

We would like to set up a SIP trunk and connect to Fusion PBX behind a PF Sense Firewall. We would like to configure incoming and outgoing calling.

...save power. Today, Flexisip supports native push systems of Android, iOS and Windows Phone ; but can also delegate the work of sending the push request to a tier service using HTTP GET Flexisip software now comprises three modules: proxy, presence and conference (the latter being required for Linphone's group chat features). This server suite is typically suitable to deploy your own cloud SIP service tuned for your Linphone-based application, especially since it supports push notifications (including iOS13 new requirements) Push Notifications PushNotification module allows Flexisip to wake a liblinphone-basd application up when a chat message or call invite cannot be delivered because the application is unavailable. This feature has become crucial since mobile OSs got used...

I need help finding a free or low cost app that I can run a virtual number through so my virtual assistant can answer calls. Any help would be appreciated and I can pay

Install new certificate on Windows Tomcat server - got cert errors Certificate does not match name Subject IT Valid from 14/Jan/2021 to 14/Apr/2021 Issuer IT TLS Certificate is not trusted The certificate is not signed by a trusted authority (checking against Mozilla's root store). If you bought the certificate from a trusted authority, you probably just need to install one or more Intermediate certificates. Contact your certificate provider for assistance doing this for your server platform.

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.. you would be combining and into 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

...solution must be server friendly, we prefers it to be linux/web oriented but won't mind if its windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better). 2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple callers id which to be round-robinly used for outgoing calls, if caller-id is defined for the sip-account then will only send calls using that caller-id instead. 3. We would be able to define "What to do, when a called is Picked" means, if a call is connected, we would be able to define if the system is t...

...Cisco CUBE. My SIP provider is sending the INVITE package with wrong sip number as you can see bellow: Received: INVITE sip:765617@ SIP/2.0 (should be 0893540111) v:SIP/2.0/UDP ;rport;branch=z9hG4bK7cvF8Zyv4U9yD Route:<sip:765617@> Max-Forwards:65 f:<sip:0477770613@>;tag=4rDev7jZe34jH t:<sip:0893540111@sip> i:9f452181-e17c-1239-eabe-00163ea67143 CSeq:31664763 INVITE I've been trying to apply Conditional SIP Profile to change this header with the TO number listed bellow. (DOC I've been using: ) request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:u01@1" request INVITE sip-header To copy "sip:(.*)@" u01