Sip softphone symbian jobs

This project is for development of HTML, CSS and JavaScript to implement the user interface for a Chrome extension for a VOIP softphone. Phone functions (making and receiving calls, etc) are not part of the development scope. Only the UI needs to be built; connecting to the actual phone system will be done separately. Javascript to be written in plain JS without frameworks such as React/Angular/etc, you must be an expert in DOM manipulation without use of frameworks or helper libraries. Design will be provided as an Adobe XD file, and we have a specification document for how the Javascript interface will work. Please show me your most similar project you have developed bidding

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

FREELANCE ARTIST WANTED! TAFE NSW Event students are holding a livestream paint &sip in partnership with Youth Off The Street. We are looking for a painter who is comfortable leading a class online, and can supply their own materials. (excluding canvas) You must be available on November 4th from 4pm onwards. You will be on location with a set design and layout for your convenience. Amazing opportunity to reach a large audience across Sydney and be a part of a charitable event. *We hope you can offer your services out of goodwill. Otherwise, a sponsorship package can be arranged.

iam looking for an android app which is similair to silver dialer/i tel dialer (referrence ) which is compatible with g729 audio codec & works with all standard soft switches. works well with minimal bandwidth

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

We are looking fo...separated in two parts. One is mobile application 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 will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This 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 to include option to show bal...

We are trying to build asterisk system based on the following requirement. Asterisk (R) and Asterisk (B) Asterisk (R) which is Raspberry PI based and attached with USB dongle with no SIM card Asterisk (B) is connected to some SIM box but with SIM card attached Using softphone, we are connected to Asterisk (R) or Asterisk (B) and make a call to some number. Asterisk (B) will send some code to Asterisk (R) to remote link up the cell / mobile signal via IP to initiated the call Both Asterisk (R) and (B) are ready, and your job is to link up the (R) and (B) so (B) can link up cell / mobile signal remotely

... up and working and proxying to SIP server. i have built out but i might be missing something so small need someone to help troubleshoot to get it working.. i have 1.) use or hopefully to verify webrtc client demo (login with sip:1111 @ with wss:// ) SIP PBX <---->Kamailio <----> webrtc client () the config im using is this basic need = i need to use webrtc to register as sip and make a call like sip (zoiper on the web) after research i need to do this with WSS:// since our PBX is only SIP this is the route im going so im trying

Hello, would you be able to help us provisionning some equipment (SIP phones and routers) on GenieACS 1.2 ?

We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gateway ...

Need a business logo to represent a fun creative art class studio. Where you can sip a beverage while being creative painting Business Name: Harmony Art Tagline with Business Name: Sip, Paint, Create

Need a business logo for an art studio offering sip n paint fun art classes. 1. 1st logo to just read: Harmony Arr 2. 2nd logo as above but slso incclude tagline Sip n Paint

i want to convert my android phone as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. (VOIP CALL TERMINATION) so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723

I need an Calling Android app. for VOIP Business like Itel Dialer with features of register with email and recharge or with Pin Sip by company

mini POC WEBRTC-to-SIP architecture backend buildout for SIP server to webrtc (kamailio or janus) docker build setup to Sample this SIP-WEBRTC (to enable websockets) client= webrtc sip register=SIP server does RTP have to flow through kamailio or SIP-WEBRTC as well trying to model off this

Need a call to provide a challenge on Twilio sip trunking solution.

hello, we would like to create a web softphone for our asterisk freepbx server. the page must have username and password to access the web phone. on the same page there must be information such as, campaigns enabled, calls waiting, calls answered, and average conversation time. all on one page. then we would like to have a control panel where to enable users to the various campaigns / queues

Hi, I need you to configure opensips on my ubuntu server, it has to support calls from linphone to linphone, sip messages and new users creation.

we would like to have both the sip client and the telephone bar where the user sees the code where it is inserted and where there is the possibility to pause the sip

...looking to develop a program that dynamically creates Syllogism questions based on a series of logical rules. A syllogism is a logical statement describing connections between individuals and categories, namely between a subject and a predicate. There are four main types of syllogistic statements: "Every S is P", abbreviated as SaP "No S is P", abbreviated as SeP "Some S is P", abbreviated as SiP "Some S is not P", abbreviated as SoP Here, the lowercase letter between the subject and the predicate represents the type of the syllogism. A syllogism can be inferred from two other syllogisms that contain either the subject or the predicate, called respectively the minor premise and the major premise, based on 24 different rules of inference...

I’m looking for an Android/iOS app developer with voip experience to customize an Android open source xmpp client Requirements: Integrate Jingle/SIP Add a dial key pad Add payment gateway Some graphical modifications (Logo, BG color, Shape baloom message etc. )

I need simple SIP client works from command line working on tcp Transport to make a calls: the app inputs must be like this:: -u username -p password -d domain -c called_number -t TCPUPD programing language python or .net c#

I am looking for someone to develop a logo for an application that we are developing called My Global Phone. The brand is a consumer-based product that provides affordable phone solutions such as: Phone Number Parking / Number Forwarding / Global Virtual Numbers / SIP Trunking Since this will be part of a mobile application as well I am thinking a pictorial mark / logo symbol combined with a wordmark. The ability to invert the colour between a light and a dark theme would be preferable. For example black text which can be inverted to white. My preference is a minimalist type logo. Need the source high quality files and copyright.

...the dream of our father alive by naming our brand, 9 Orphans Distillery. We will predominantly produce Gin and Rum. Gin: A perfect drink for a hot summers day or to satisfy a gin lover's need for a special occasion, gift or just another Friday night. Enjoyable on its own or as part of a cocktail. This gin made from a blend of botanicals that includes juniper, buchu, lavender and lemon peel. Each sip begins with a floral bouquet leading to a lasting and clean buchu finish. The lemon peel giving a lingering mouthfeel that would leave you wanting more. Rum: Breaking all miss conceived stereotypes that is rum, but not forgetting its roots. This rum is great for sipping on the rocks, neat or enjoyed with a dash of lime or part of a cocktail. The crafting process excludes any fla...

I have a voip project need use the LinphoneSDK + Flexisip server. I have setup the Flexisip server but I found if we use account. Everything works smoothly. call from LTE to Wifi. Works fine. But switch to the sip server we created. It only work with in the same local network. No matter we enable the ice/stun or not. I think there some configuration thing I miss. also there one more thing about the mysql database authentication.I try to enable the mysql database authentication. But it not work. The file storeage authentication works fine. I need some developer with the experience about the LinphoneSDK + Flexisip server to help me. Thanks Frank

Kamailio Server setup on ubuntu or Centos connected with linphone sip client .use UDP Protocol

Hi Ibrahim Ali I need setup sip account on 3CX and i need send the traffic to it from my soft switch setup will be VIA Team viewer

I need setup sip account on 3CX and i need send the traffic to it from my soft switch setup will be VIA Team iewer

I would like to configure flexisip sever with push notifications along with asterisk and any softswitch. Prospect freelancers should have proven experience with SIP and Asterisk and ability to complete job on time.

Configuration of the Cisco 819 Modem & VPN Link 1) Configuration of the sim with the GSM Operator 2) Config of Main and Backup Sim in case of the Carrier Failure. 3) Setup a Router in a remote site that has Public Ip with Port. 4) Make sure traffic reaches through the VP...Port. 4) Make sure traffic reaches through the VPN from Remote Side via Public Ip and Port. 5) Assign a Local Ip based on the VPN Traffic . from Remote to Cisco 819 Modem. 6) Prepare complete Document and COngiration steps and document for our record. 7) reset and factory default setup to ensure configuration remain there. 8) Configure the GOIP Device as well. 9) Connect the GOIP device with SIP Server to make calls. we have lcoal and Cloud both Server to make sure GOIP Is Connected to CLoud is impo...

Hi 1- Fix odoo softphone and add udp connection to it if possible . 2- Create calls manager module to connect to Issabel pbx server through AMI ( Asterisk Manager Interface ) .Or edit existing module but with direct connection without using any middleware software like odoopbx for example .I have pdf file also for requirements .

Hi 1- Fix odoo softphone and add udp connection to it if possible . 2- Create calls manager module to connect to Issabel pbx server through AMI ( Asterisk Manager Interface ) .Or edit existing module but with direct connection without using any middleware software like odoopbx for example .I have pdf file also for requirements .

Signalwire's SIP trunk on asterisk

We are currently looking for experienced Call Center technicians and developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

I am currently looking for experienced Call Center technicians and developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/pr...

We are currently looking for experienced Call Center technicians and developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

We are currently looking for experienced Call Center technicians and developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

We are looking for someone with experience to suggest and set up a PBX. We need the PBX to have the following features. - Easy to manage with a graphical interface - Should have a built in firewall and fail2ban support - It needs to support multiple sip domains - No Limits on the number of users, extensions, Hunt group or Auto-attendant. - Automatic backup to another instance on the cloud. - Need to be able to automatically switch to the backup in case of failure. We need the freelancer to suggest a PBX (open source if possible), install configure and document the read the description carefully before bidding.

I need you to recommend me a call center platform. I need a script upload that is easy to use along with a softphone hooked to the account

Telecommunications Consulting - Implementation of SIP Platforms, traditional PBX, Data Network Implementation. Technology Consulting Service Desk Team Outsourcing Sale of Technology Equipment (Networks, Computers, Servers)

extend our existing SIP server implementation based on JAVA as an eclipse RCP plugin with full SIP functionality

Hi Its new Channel related to Finance, and today I want to have video on SIP ( systematic Investment Plan ). Need to explain what is SIP and How it will help you to create wealth for you.

Hello Friends, I need help from someone experienced in JITSI. I'll provide you AWS Cloud instance access. and you have to install and configure jitsi on AWS with the following features. 1. eatherpad. 2. JIGASI (SIP and CC features with Google Translate API ). 3. JIBRI 3. SSL All are builtin with jitsi you just have to install and configure them properly. Budget is around $50-$60 (USD) No Time waster please. Bid only you have knowledge on JITSI Meet Configuration with those. Thanks, Ranjit

I need someone to build me a simple Web Application using React and Node. I would like the this app to be able to register with my asterisk server through the browser. The required fields will be: - Username - Password - Host - Port Technologies that can be used to help accomplish this are: jssip (npm react-sip), ml5sip, or janus. I need this app to be organized and simple as it will be used for learning. I can provide the credentials for testing if it connects to my asterisk server. App should be able to register, make call, and hangup. This will require WebRTC. No need to worry about trunking as my asterisk server will take care of that. See screenshots for designs.

hi i installed goautodial. All working, sip trung registred. But i just got a problem when i try to mu a call: asterisk log : [Jul 16 20:35:28] WARNING1855[C-00000006]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"M7162035280000000001" <sip:5164536886@************:5070>;tag=as346c9066' [Jul 16 20:35:28] ERROR2774[C-00000005]: member.c:389 member_exec: unable to answer call [Jul 16 20:35:28] WARNING2774[C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel if someone help me to make a call for french phone. Thank you.

I require a SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and freepbx. I need a SIP & Jitsi expert for: 1) create a sip/h323 connector for jitsi, 2) help me with getting an affordable india phone number for inbound connections – Twilio pricing is not affordable for Indian numbers 3) setting of sip trunk between jitsi and freepbx, so the participants from pstn can connect to the jitsi instance 4) Connect our jitsi instance to google transcription. I will provide the API key

Need a writer for sip trunking website to write 1-2 articles and web content every week of 500 words each.

is an example of a avatar that is speaking, but it is limited to the consumer answering not with voice but by typing on video. We want to allow the consumer to speak in to video, we know we can do the speech to text, as we already do that on Telephone so we can hook in via sip to listen and then make the bot answer, but we don't know how to get this avatar character in as a video participant, allowing us to control the lip movement we do via api. So we thought maybe a screen share, where it is a website which auto plays the avatar or if not, we would have to send over mp4 video files to the solution for each prompt the avatar says. see this type in a question and see how it responds, we want this same thing, but not have