Softclient sip trabalhos

Outbound: -Realtime Monitor - here we can see the agents realtime -Scoreboard - Overview of top 10 agents -Manage Dialer - Here i can give a boost to dialer & and the list what i have imported, i can choos to call randomize or begin with the oldest data or with the new imported data etc. -Manage Resultcodes - I can add here a resultcode to any projects what i have selected. -Search Record...can see here the whole time they logged in -Agent Hours - i can see which days the agent has logged in -Resultcodes Report - here i can see between date filters and projects (each projects has different resultcodes) and see here like 77 voicemails 120 not interested -CDR -Batch Report - here i can see how good the imported data is. Ex. data A) 20 sales Data B) 10 sales -Agent Report Settings: -...

Voip application developer ios and android

We are looking for solution lik...will be separated in two is mobile applications and another one is registration server. The mobile application will register to a server and accept call from that server with IXA OR SIP protocol. After that call terminate to GSM Network. (this part like traditional gsm gateway). This mobile application will work only wifi internet connectivity. Beacuse gsm intenret date normally disbale during any gsm call. All call will pass through gsm 729 codec. The registration server may be voip switch or asterisk or any other server. The server will receive call from another voip switch server with sip protocol. Certain number of registered mobile will be able to assign in a group of gateway. On server have have include option show balance throu...

...experienced in managing deadlines. Our company is looking for someone to understand our views and can participate in the race to reach the goal. Vacancy 1 Responsibility: 1. Architect and solution engineer to develop Video Collaboration platforms using SIP and webRTC. 2. REST API based integration as a service development for Applications that need to interact in real-time with each other. 3. Develop technology use cases, architect overall solution, engineer implementation on middleware and 3rd party systems using standards such as webRTC, SIP Video interop, H264 based video conferencing platforms etc. Business requirements and product management support would be provided. 4. Develop sustainable, scalable interface with user experience use case support for product develo...

There is a limitation with SIP service in CheckPoint. Freelancer is expected to make a Service object with the same port to fix RTP packet issue with Checkpoint 730. It requires some NAT, Firewall rules to fix the call drop / 1-way audio muted issue. Experienced Checkpoint engineers only.

...installation. Our customer has a CRM used by call centre consultants and hosted at a different cloud provider. All telephony based activity is done via the CRM (no physical phones or 3rd party soft phones) to connect to the FreePBX server. The customer has a implementation, using WebRTC. The implementation is making use of 2 secure connections (wss://); 1. to handle the voice and SIP 2. to handle server requests such as Login, make call etc. and receive call progress data such as channel added / removed, connected An important feature for the call centre is that all sales calls are starting as a conference, this to create a customer experience that allows for adding additional persons without the unpleasant silence etc. associated with being put into a conference room. An AMI

...------------------------------------------ 1. Brand name needs to be Single or Multiple words (Example -Single Word: Google, Multiple words: Facebook) 2. Name needs to be timeless, tireless, easy to say and remember  3. No other language except - English.  (Example: The word "Chai" will not be considered as English) 4. Do Not Use the following words on the Brand Name -CAFE, TEA, COFFEE, JUICE, CUP, SIP, BEAN, BREW (Example: Brand Names like Cafe Time, Tea World, Juice Center , Coffee Palace will be REJECTED) 5. Avoid Plagiarism and Avoid duplicate entries....

Hi debjeetfailure6, I noticed your profile and would like to offer you my project. We can discuss any details over chat. Wir suchen nach jemandem, der uns bei unserer bestehenden Jitsi Konfiguration noch eine Einwahl via SIP ermöglicht. PIN Eingabe via Telefon -> Übergabe Conference ID und Join in den jeweiligen Raum. VPS mit Jitsi-Meet ist vorhanden, ein Sipgate Basic Account für die Einwahl auch. Dieser müsste zum Testen reichen, falls danach was anderes notwendig ist, könnte auch ein easybell Trunk gemietet werden etc. Was wäre der Zeitaufwand dafür in etwa das IVR zu generieren etc.?

need a good android sip developer. with knowledge of native libarary, dagger2 and rxjava. I need developer for full time with expirence is 2 year. I can pay salary milestone on weekly or monthly what ever is you prefered. 20-21K fixed price. type 3rd word is andscope. so, I understand you read my job.

hello i have a raspberry pi that have Rasterisk on it i am using chan_dongle solution to use Huawei E160 for a GSM port my end is this : Port1 is talking B, during this time a sip call(from C to D) is coming to Asterisk , Asterisk should hold call on port1 and B , and port1 should call D when call connected to D , port1 should merge all calls (Conference) it means now port , B , D , C is on one call

Hello, I'm looking for a VOIP professional that can build a cloud app that will analyze incoming calls in real time, and get the SIP Address from SIP Header. The purpose of this tool is to help protect against DDoS attacks on phone numbers. Requirements: 1) Analyze incoming calls in real time, and get the SIP Address from SIP Header. 2) Block calls or send to CAPTCHA if a trigger has been activated, For example: 5 simultaneously calls from the same SIP Address. 3) Will discuss further with the proper candidate . If you will apply to this project without reading the content, your request will be deleted immediately!

Need help setting up a Cisco Unified Communications Manager with several phone numbers and sip trunks, as well as configuring the VLANS on my switch

I have 2 phones cisco ip phone 7970. they are both stuck in a boot loop trying to upgrade firmware. i'd like to fix the firmware problem and setup the phones to use at my business location through the service. possibly with a virtual pbx.

...About; Settings Invite to Use; Privacy). The app should integrate with iTelswitch. The main functions of the app will include: 1. Making & Receiving VoIP Calls Over 3G, WiFi, GPRS 2. Should not kill the battery of the mobile device 3. Address Book Integration with local Phone Address Book AND the platform 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) 7. Media Support RTP (Real Time Transport Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Pac...

We want to integrate asterisk free PBX with Avaya and siemens PBX for one of our customer... If any one local KSA available let us know..

We need crm for sales and leads with telephone system sip integration , i find bitrix24 if someone can customize it.

Maintain customer details Login for clients to see there investment details Login for lead providers to see there leads details and amount they...they received/ pending from us Login for Admin to see complete details Maintaining Customer's Investment details Sending Renewal notice to them Sending seasonal greetings to all Maintaining Mutual Portfolio with folio update option Simple website to display all of my products with update features so few products can be updated on regular basis from our end Few useful tools [calculators] [i.e. SIP/ SWP returns. Loan calc etc..] Domain we will purchase separately to keep it in our control Note: Other features can be discussed before finalizing We deal with all types of investment products: INSURANCE, INVESTMENT, REAL ESTATE, LOANS...

We need to install asterisk to the debian 10 server. We need to start the asterisk on server, add 2 SIP accounts and organize SIP calls between them.

We have a online sip & paint event and need artist to instruct adults and children

Hello. We are triying to work for home and need to configure our VOIP System for this. Current System: - VOIP Grandstream Server UCm6202. - 2 x IP Grandstream Phones already configured, SIP accounts ready. Basically everything works in the local LAN. - Edgerouters in both sides. --- Option 1 Requirements: IPSEC Site to Site Tunnel Description: Connect the phones of external LANS trough an IPSEC tunnel. We already have an IPSEC tunnel running under our ubiquiti edgerouters. It works allright, and we can currently ping and access other kind services from one LAN to the other (CCTV, webservers, etc).. Howemever we cannot ping the Grandstream server from the outside LAN, so the IP Phones cannot connect. Seems like its blocking IPs from other subnets. Work to do: Configure server t...

We want to configure flexisip server, help us, how we shall configure SIP server with TLS / TCP port in Linphone application with media encyption DTLS.

Strictly bid if you are familiar with softphone/webrtc and web app Works on major web browsers Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat)

I have AWS instant hosting PBX, i would like to authenticate SIP trunk using IP Base. Is there any chance you can assist me. thank you ahead.

Paint and Sip startup is looking for someone to do the search engine optimization. We are a little business, engaged in DIY, we would like to be involved in the process (eg. keywords set up). Ohh and our website is:

Looking to transitt SIP-services to Twillio, please response with earlier experience and project using Twillio.

I will have the system set up and hardware connected. What I need is someone who can do: 2. Codec will be standalone and not registered to an Call Control (CUCM, WEbex) a. Needs to be able to join others’ Webex, Zooom, etc conferences by dialing the SIP URI, etc. b. Needs to be able to do point 2 point ad-hoc conferences with other Codecs. Encrypted video/audio. Possible Side requirement to integrate with Alexa? Alexa currently controls the Samsung hub that controls the lights in the room.

Bid if you are familiar with softphone/webrtc and web app Works on major web browsers Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat)

Configuracion de una conexion STM1 con el Operador de Comunicaciones para conectarse a la red de telefonia, y configurar un troncal SIP Voip para poder hacer llamadas telefonicas hacia esa conexion, el equipo a configurar es un Cisco ASR1000 series.

Hi Friend, My mini project i need source code to config integrate Opensource PJSIP using WPF c#. Many tutorial on Github but i duno why im not success build Dll from pjsip project and integrate it into c# project. The project result: - Guide to build PJSIP to DLL using in c# ( i try here but not success ) ...pjsip project and integrate it into c# project. The project result: - Guide to build PJSIP to DLL using in c# ( i try here but not success ) - WPF project import PJSIP DLL/ Wrapper C++ function by Swig to call in c# - Can call function support by PJSIP in c# - Can register and make call from WPF Version pjsip is newest on The SIP infor account i will send later

# Background We are the Software Company involving in RealTime Communication like WebRTC, SIP, HLS, DASH. I am looking for the great developer who can write very clean code and learn new things very quickly. # Requirement Fullstack engineers who can solve any problem about coding and communication issue in your sides For this job, we are hiring freelancer only, don't need developers in big agency # Question You must answer questions when you apply to this job. If not, I will reject you immediately 1. When creating new React.js Front Side project, what tools and libraries do you use and why(for example, CSS library, Middleware library)? 2. This is my own sample code for react-native project. Can you suggest how I can improve

Looking for a Mac OS Developer, with experience in VoIP and PJSIP Library. Need to build a custom Softphone that can work with our SIP Servers. Here will be the Flow: Application will call an API with username and password to get the SIP Credential. Those credentials will be used to configure Softphone to register on phone system. User should be able to Choose Codecs, Add multiple SIP Accounts. We will hire only people/company who has an experience in VoIP and Sogtphone Application for Mac. Must have worked on similar project and must provide reference. Here you can find windows version.

We are a new VOIP and PBX provider and we are looking for a softphone which has all features thts in BRIA and XLITE and can be used in Windows, Mac, Android and iOS devices. Plz place your bid only if you have developed similar application. To award you the project i would need to see the demo of your application.

Hi i need a Flexisip Server Setup & Configuration complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications. I need good service. when install complete then i need complete installing document

Hi i need a Flexisip Server Setup & Configuration server to be set up, complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications. when install complete then i need complete installing document

I need you to develop some software for me. I would like this software to be developed for Linux. I need asterisks downloaded and set up on my server. I would like my software to perform following actions; - auto Call - text to speech, with access to change script - dtmf - record voice - send results to telegram

We have an asterisk server at our HQ and we have sip phones in our branch offices. We want to Develop a SIP softphone that we customize with our brand and users can install in their windows, mac, ipad, android etc....We need to do voice and video calls with the softphone.....something like LinePhone and Zoiper.

hello, we are looking for cgrates expert developer who can develop cgrates with our existing open sip server. only experts are welcome on this project. Thanks

I have a VPS server that is set up to connect to a few VOIP providers. One of my VOIP providers recently changed the SIP and I need to update my VPS along with the user/pw permissions. My VPS is running: Ubuntu 16.04 (64 bits) I need this done QUICKLY.

1) We need developer for softphone client for Windows, Linux and Mac OS X with SIP and IAX capabilities. IAX and SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure. Similar product available is Zoiper (2) We require application for management of PBX customer. We are currently using Issabel. In this application, we require the management under (a) Super User i.e. company and (b) User: Business Partner. Extension and Phone configuration i.e. users creation can be managed by this application as we do in Hardcore PBX software: Issabel. Activation and deactivation, renewals can be managed through this application. Funds allocation for management of user creation and renewal is to be there in application.

I need both ios and android app made. a sip client i can connect to any sip server and use username and password (example apps like linphone). I also need DTMF signals to be received. Also need to enable input of sound as microphone / bluetooth / or pre recorded mp3 files - using a browse function and play and pause

Hi there, Hope you are doing well. I noticed that you have previously placed a bid to rebrand linphone. I would like to do the same for Windows and Mac OS client with our company brand and linked to our SIP server. Can you please provide an estimate and timeline for such a job please?

...produktu na rynek we współpracy z odpowiednimi działami w organizacji (R&D, Technical Support, Sales); • Współpracujesz z kluczowymi dostawcami oraz klientami firmy; Wymagane kompetencje: • Masz wyższe wykształcenie techniczne (preferowane elektronika, informatyka, telekomunikacja, elektrotechnika, automatyka); • Masz praktyczne doświadczenia z sieciami IP do aplikacji telekomunikacyjnych (m.in. VOiP, SIP, Asterisk, Trixbox, SNMP); • Komunikujesz się w języku angielskim; • Masz silną orientację na realizację celów; • Jesteś kreatywny, dobrze zorganizowany, samodzielny, jesteś nastawiony na współpracę; Nice to have: • Doświadczenie w pracy jako System Engineer / Product Owner; • Doświadczenie w pracy w meto...

...I've tried this on both iphone and android client. Can any asterisk/freepbx professional provide me a quick solution for a fee? Full log: -- SIP/6566811234-00000005 answered PJSIP/90101-00000005 -- Channel SIP/6566811234-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> -- Channel PJSIP/90101-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> [2020-04-05 13:30:53] WARNING[19800]: res_http_websocket.c:508 ws_safe_read: Web socket closed abruptly -- Removed contact 'sip:1se6p552@;transport=ws' from AOR '90101' due to shutdown == Contact 90101/sip:1se6p552@;transport=ws has been deleted [2020-04-05 13:30:54] WARNI...

I need an Android app. I would like it designed and built. I want to implement PJSIP native library and build function for sending sms and calling using voip server

I did already install VoIP system with 5 nodes (please, see detail in attached file) - 02 nodes: dSIProuter (Kamailio, RTPengine, Mysql, Nginx (docker)) - 02 nodes FusionPBX (Did config HA with Postgresql BDR & HAproxy) - 01 node install Homer (For monitoring the SIP, RTP & WebRTC) I need a expert to advise, fix some bugs and optimize this entire pilot platform. Include the following: - Advice on system structure. - Config the WEBRTC to SIP client and server (WS on Kamailio, RTP) - Install TURN server - Configure dSIProuter Cluster (HA) and can load balance traffic to FusionPBX nodes

This task is straight forward. I need a iOS linphone app that recieve VoIP push and work in background. Only bid if you have done this before. No time waster. I will need to see evidence you have done this or we start with a proof of concept. Open to other iOS mobile sip client that can work with freeswitch and recieve push. I will provide server detail - i use fusionpbx

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive...

Link Phone Trunk to Big blue button Big blue button is installed i need you to create a connection to my German sip provider

Hello, I am looking for an expert developer with WebRTC XMPP SIP android and ios app developer and integrating with Raspberry Pi. With 90days Recording feature all the resources required for building the project will be provided.

...increase performance (high availability, low latency) and automation (service delivery, testing...) * Design and management of our SIP interconnections with operators across Europe. * Voice quality monitoring and troubleshooting * Development of internal tools to automate / offer web self-service for most common tasks You are : Passionate about technology and expert in VoIP, you wish to join a fast-paced company and a technical team strongly oriented towards open source projects. As an innovative person by nature, you are eager to learn new technologies and to take up new challenges. You master most of the following technologies and methods : * VoIP protocols (SIP, RTP, WebRTC) * Analysis and investigation (logs, IP frames, pcap traces...) * Shell and Python scripting * ...