Symbian sip softphonetrabajos
Our product is described as a partner (chaser) for tequila, mezcal and/or beer. It consists in a medley of flavors that bring out a flavorful concentration of strong, sweet, aromatic spiciness to the drink in question. You take one sip of mezcal and then some of our product to enjoy. The color is dark red, and you can see the pepper floating around in a transparent cristal bottle. We use only natural flavors, such as: orange juice, tomato, lime, and spicy sauces. We need a label, for the exterior front of the bottle, as well as the list of ingredients on the back side, we want a modern looking, innovative, artsy, crafty, design. We need you to consider that due to recent regulations, there will be at least two big hexagons that cover part of the lable (top right). No mascots or cha...
...groups for file access – As existing – see attachment plus Chltd users for Chltd client files and admin In office working – via wired network with Panasonic telephone system with PoE connections apart from GH Surface linke3d through Wifi Sonic Wall – Hardware Firewall BT Whole Home Wireless Service for visitors (set up in DMZ) Remote working via Netextender VPN (Sonic Wall) and using Panasonic Softphone Sophos Central for Endpoint Security Cloudberry/Amazon S3 offsite backups Zen internet ISP Most remote users use RDP to connect to their desktops which are kept powered up There are five printers/scanners in the office to be shared by all users Longstanding Issues to resolve It would be good to configure Wake on Lan on all office -base desktops but this h...
We have a new Cisco UC560 that needs configuration. Device - Cisco UC 560 Needs to configure 1 SIP trunk and 5 - 10 Extensions Phase 1 -Getting the SIP trunk up -Getting the Extensions configured -Internal calls between extensions to be working perfectly Phase 2 -Ring groups -External calls incoming and outgoing to be configured -External caller IDs to be configured -Call transfers to work fine Phase 3 -Small configuration touchups - after 7 days
...page in a jpg and pdf format. The SIZE of each Promo Voucher will be 3 5/8 x 8.5 Use one sold background so I can cut it into three vouchers and get a bleed. I attached a sample called 'Bleed and Cut sample'. Attached are the colors of the boat - neon green and bright blue FRONT: Use the Term Boarding Pass in larger print Boarding Pass - Sample attached called Boarding Pass Attached is my SIP-n-Cycle Pedal Cruise logo. On the perforated side I need the: Promo Voucher' information To: From: Message: Promo Code Promo Value BACK: Private Parties Birthday Parties Corporate Team Building Family Reunions bachelor/bachelorette Parties BOOK NOW at 334.399.2387 Small print on the bottom: Not redeemable for cash Good until 12/31/2020 Can not be combined with
Adding some codecs to existing SIP Phone project (Java)
We are looking to customise Bitrix24 to be used for a call centre. There are 2 ways our team will use the software. 1. Each person will have a set of leads to call on a regular basis. 2. We have campaigns where leads are uploaded and each call is then handed to an agent. The pr...person will have a set of leads to call on a regular basis. 2. We have campaigns where leads are uploaded and each call is then handed to an agent. The process is outbound. We need call recording and history for each client / contact recorded on the system. There will be custom fields for data entry for each new campaign. Daily / Weekly and Monthly reporting needed. Integrate our VoIP / SIP channels to dial out from software If you have other suggestions for software then we are open to loo...
I need some calculator on my wordpress site....( SIP CALCULETOR, LUMPSUM CALCULETOR, FD CALCULETOR , PPF CALCULETOR, INCOME TAX CALCULETO )Only bid if you can do the job.
I have my own auto dailer astrisk but i fase issue i set call limit 3 or 4 min duration but all calls ended with 120 sec this my agi code if u know how to solve please contact me mysqli_query($conn,"UPDATE `static` SET `number`='".$calledNumber."' , ...FROM `products` WHERE `product_id`='".$row_trunk['product_id']."' LIMIT 1"); $row_products = mysqli_fetch_assoc($result_products); $agi->set_variable("trunk_id", $row_trunk['product_id']); $agi->exec('Set', "CALLERID(all)=$calledNumber"); $statusDial = $agi->exec_dial('SIP',$row_trunk[�...
We are looking for a C++ Senior Developer who is very good expertise in PJ SIP. We are porting our existing SIP applications from using the Radvision SIP stack to using PJ SIP. We are looking for experienced C++ developers that have worked with PJ SIP. The applicant must know SIP and work in-depth with PJ SIP The customer develops a soft-switch that provides, prepaid, Class4, and Class5 services. These are our SIP Proxy, SIP Registra, and the SIP Call Processing layers All the application run on RHEL/CentOS 7 and are multi-threaded Candidate must have at least 5+ years of experience. Candidate must have good communication skills, as he has to talk to US-based Clients. The candidate has no problem in Working Night Shi...
Hello I have on AWS instance a freePBX server installed, and sip trunk is twilio. I installed Zoiper on my android phone. I am willing to install other apps. If you need to login to server I prefer you use TeamViewer. :) I'd like to have somebody(YOU) to help me configure a softphone app on my cellphone to start making phone calls from it. Please reply with a timeframe this task would take you
Looking at additional developers that are familiar with codebase of ionic cordova/capacitor building apps/ideas in Android/IOS with webRTC SIP
Following Features Required:- 1. Answering Rules 2. Auto-Attendant and IVR 3. Call Monitoring 4. Call Recording 5. Call Reporting 6. Hunt Groups 7. Intelligent Call Routing 8. Voicemail to Email Delivery Guide us to change sip settings. Full access required after installation and setup.
I need an Android and iOS app. I would like it designed and built. App will contain: 1/ Splash screen (logotype+loading...) 2/ Login (register) screen 3/ Main screen with navigation between: a/ Keypad b/ Contacts c/ Recent (call history) 4/ Call in progress screen with ability: a/ display call duration b/ hold/unhold c/ disable/enable speaker d/ disable/enable mic e/ show/hide keypad 5/ display notifications when received incoming call/message
- Export embedded Android9.0 - SIP application - Optimize WIFI/Bluetooth(11r and SIP) - Optimize BSP(GPIO) and related the application - knowledge ARM Cortex-A - knowlesage Audio Note; It does not need H.264 codec etc.. Audio only
I have to collect call log from telnyx sip trunk to my website which should be done by .net framework 4.0 only (asp.net c# only)
Full means: - a WORKING .ino - accept inbound call, play wav/ogg (answering machine) - record inbound audio - make call and play wav/ogg on answering ULaw, C++ —> FLUENT English, do NOT think you can copy some existing example from the internet and simply cash. FIXED PRICE $ 250.01 I only read EXACT same amount bids, everything else is considered automatic bid and immediately deleted unread. As soon as you start negotiating, I’m done too.
Hi, We want to enable dialing into our Jitsi server from a Cisco telepresence hardware. We are able to setup jitsi to accept connection from SIP server but do not have experience from the telepresence hardware to the SIP server. Due to lack of documentation regarding the SIP server side, my understanding could be slightly wrong. Hence I am attaching a flow chart of what we think is the correct flow
You will be supplied with two training videos that will be a s...palm trees blowing in the light breeze, you will then fade into our tutorial video. Then after our tutorial video ends you will transition into the animation again this time with the male per description above, he will walk onto the screen with the same island background but their will be a beach chair and table with a cocktail on it, he will lay down down on the chair and cross his feet and take a sip of the cocktail then fade out. Can I have a price for these two animations including fading in and out of our tutorial. Not when there is walking onto the screen we would want the flippers flicking on their feet. We will provide a full script so you are aware of when and where the animations will go with the two traini...
You will be supplied with two training videos that will be a ...palm trees blowing in the light breeze, you will then fade into our tutorial video. Then after our tutorial video ends you will transition into the animation again this time with the male per description above, he will walk onto the screen with the same island background but their will be a beach chair and table with a cocktail on it, he will lay down down on the chair and cross his feet and take a sip of the cocktail then fade out. Can I have a price for these two animations including fading in and out of our tutorial. Not when there is walking onto the screen we would want the flippers flicking on their feet. We will provide a full script so you are aware of when and where the animations will go with the two trainin...
We have created an app which is based on SIP calling feature in Mac system. In this case I need to stream call audio to our server for eg. Microphone and speaker. Not only call audio if any audio Mac system is producing through speaker (User can use headphone) for eg. Youtube, you need to capture in buffer. You need to implement audio feature in C or C++ and provide one interface for swift language in Xcode. For Understanding you can try like this, You can start meeting from zoom, slack or Skype etc and in the same time you can run the your application where you can capture meeting call audio from mic and speaker (Try with Headphone) and then capture the buffer of mic and speaker. If you are able to do that you need to create this feature in C or C++ language and provide an interfa...
Hi, I am having one dedicated server where I hosted the vicidial server. I am done the necessary configuration. When I register the carrier settings, I am facing the SIP account is not registered and carrier settings is new to me. I am using the Webphone and it is also not registering. It needs to be resolved.
SBC is active and operational all sip will be done on inside, no networking will be done by asterisk engineer sip trunks in and out Auto Attendant set up and recording handsets and users set up voicemail set up explanation of hunt group setup for round robin and collective overall complete system setup of Asterisk Issabel, remote technician available for datacenter and onsite for login in phones. Isabel already installed and tested with one handset. all handsets grand stream
Brand Name: Half Glass Haydn Brand Story: The owner of the brand is known as a “the wine thief” as he can’t resist to take a sip of someone else’s glass when they are not looking. Guest and sometimes even random people often find their glass half when they left their wine unattended for a minute. He originally wanted to call the name “Wine thief”, but the name is already in use. Brand Identity: This is an entry level brand that is supposed to be tongue in cheek and quirky. A bottle of wine that is fun to buy without looking cheap. SKU’s: Dry red and Dry white Back Label He would like to have some kind of slogan on the back of the bottle. Something like “I will never trust my husband near my wine again” Colour ...
1. The sip from header will have a tech prefix before coming to kamailio 2. We will check that prefix and based on it we will chuse which diapatcher to use 3. We will remove this prefix before call and register is sent out
I have an iOS and Android app that allows users to set up a custom voicemail service. To enable the voicemail service, we identify the user's phone carri...phone number of the user, have our app reject the call, and then look for an incoming forwarded call from that user's phone number. However, it seems like there could be a simpler process. The second option I am trying to figure out is a way that I can just call the user (it's okay if the user's phone rings once or twice), and then retrieve the actual SIP header information and extract the call forwarding phone number. This way we can see at the SIP / PSTN level the number that is configured for the conditional call forwarding. To apply let me know if you know how to do this, and how long it would tak...
We, here on VMAX Digital, want to integrate our Softphone App with Push notification services. And for that, we need to setup a Flexisip push notification server.
We need a research, someone who knows well about VoIP market, provide us with a list of 50 VoIP (SIP Trunking Providers) who are easy to signup online (no need to communicate with them by mail etc.) and also they provide on their interface a full pricing list with all (A-Z) codes rates.
Hi. Do you happen to have a SIP client that runs on Arduino/esp32, both inbound and outbound? Peter
Do you happen to have a SIP client that runs on Arduino, both inbound and outbound, and working?
Embedded mobile device < develop> - relates drivers(WIFI/BT and ALSA, GPIO, LCD) - Application < Important> - It is not an application engineer for the Andoird smartphone. It is an embedded device. - SIP and WiFI/BT etc... wireless export - communicative to make asta UML - If you do not have the experience, don't contact.
We need a Unified Communications software system that can be hosted on our server at our data center that provides Secure Encrypted Communications with the following features - SIP Voice Chat Video Screen Share File Transfer Windows Mac Desktop iOS Android App
...requirements, 1. Kazoo does not have tariff need tariff in this with all type of customers. 2. Kazoo does not have Device provisioning in it for SIP devices as per SIP brands to program certain keys. Modules Required, 1. Retail Client Client for selling DIDs & Termination routing like VOIP Switch,asiign as many extension numbers to one extension,ver cool feature. to select Different trunks for different PBXs to select number of registrations per account. option must be in Retail & PBX. user interface for PBX admin & Client & Retail client. 8. Live view of SIP call with information of error. 9. SMS service like SIP simple 10. SMS delivery over DID both ways, inbound SMS & outbound SMS. Please contact only developers with the ...
I need a lightweight and WORKING version to perform inbound/outbound SIP, preferably uLaw.
we are looking to develop a plugin/ addon for chrome or firefox that can turn phone number on the PC browser into links that once clicked will call the number and ring on user SIP soft phone (bria for iphone) we will also need to be able to type a number and dial it, review current calls, review call log and have notification on PC screen of incoming call including caller id. only looking for qualified developers with proper experience please send sample of similar work you did
I need you to develop some software for me. I would like this software to be developed for Linux using PHP.
...end-to- end project planning and implementation from scoping, to activity sequencing, effort & cost estimation, risk analysis to quality management • Expertise on Contact Centre Solution (Call Centre) implementation • Designing IT/Telecom infrastructure and implementing technology to support large user groups • Comprehensive understanding of networking concepts pertaining to ISDN, TCP/IP, UDP, VOIP, SIP, H.323, Telecommunication • Effective communicator & negotiator with strong analytical, problem solving & organizational abilities • Expert levels of problem solving and analytical Ability to analyse complex, detailed structures • Good leadership skills and has the ability to guide and provide technical direction and supervision for a give...
Hello, We have a working FreePBX distro installation working since 2 years. Now we want to do following enhancement in our FreePBX server. 1. We want to use our FreePBX server for internal office text message communication. I know this can be possible using ...possible using XMPP module of FreePBX. But don't know, how to configure it. You have to do that. On the other hand we want to transfer file as well at the XMPP messaging window. I don't know whether it is possible or not using XMPP. 2. We want to do video calls between extensions. Need to do video conference as well. You have to configure our FreePBX to do so. 3. We are using Bria Professional as SIP client. You have to configure Bria with FreePBX so text messages and video calls could be done as well. Regards,...
We need a "branded" softphone with a modern design . This softphone could be based an already existing open-source project in order to let you work exclusively on "branding" and "design" of the product. The main features we need are: • sip communication, • SIP SIMPLE protocol(for instant messaging) • Branding and "modern design" • Provisioning through API • Presence status (not mandatory) • Video call (not mandatory) We already tested this two open source solutions which work with our pbx, have all features we listed before and both give open source code Linphone: Microsip:
We have two fresh installations of vicidial and goautodial. We would like to configure one to test some outbound voice blast campaigns. We would like recommendations on which product to use for our needs. So Goal is: 1- initial trunk setup to asterisk or sip trunk setup. 2- Setup 2-3 campaigns based in one file txt each with all non agent "virtual/voice blast" campaigns. We have 5 different voice messages per Campaign messages which will be based on a specific value on the list. We have text files with caller information which we can convert to csv format. We would like to have the process of uploading daily lists automatically (via cron or other interface) from a directory where daily files are transferred/deposited during the night. Also have the campaigns start automat...
Looking for a freelancer who can provide me a SIP system for outbound calls. Requirements: control over CallerID, dial through Soft phone. If such system exist, no need to construct one from zero. do not have my own server nor cloud
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...controls Enable Lobby feature Enable Recording to dropbox, Google Drive, One Drive, and local machine Enable Live streaming to youtube Enable Transcription(cc), integrated with Google Speech to text API Integrated with SIP, Telephone dial-out (India number only as of now) IVR integration Dial-in to the conference Integrate video into a react web application via iframe API Build a Chrome extension for Google and O365 Calendar Provide a Plug-In to Outlook containerize all the roles (signaling, Authentication, Media, Recording, and SIP) Customize JVB and Jibri to autoscale via Kubernetes Enable monitoring via Grafana and Prometheus Configure the infrastructure for scale and cost-effective Deploy video bridges across Geo locati...
We did SIP peer trunking between two pbx, Mypbx and Avaya. from Avaya we are able to receive calls but from mypbx we hear " all circuits are busy try later" message. kindly anyone helps me with this to solve.. regards
Hi Amal A., I noticed your profile and would like to offer you my project. We can discuss any details over chat. "Need assistance with configuration of Vicidial dialer settings and visual customisation. The server is already installed and working with carrier configured, sip registration up and calling working via configured phones in vicidial. I require assistance with configuring the dialler settings and further customisation to make the dialler & agent experience similar to the existing cloud provider I am using. Existing cloud provider is accessible through admin panel but no console. New server has admin, console & teamviewer access (Server is virtualised with low load/requirements -manual dialing/~5 agents.)"
Need assistance with configuration of Vicidial dialer settings and visual customisation. The server is already installed and working with carrier configured, sip registration up and calling working via configured phones in vicidial. I require assistance with configuring the dialler settings and further customisation to make the dialler & agent experience similar to the existing cloud provider I am using. Existing cloud provider is accessible through admin panel but no console. New server has admin, console & teamviewer access (Server is virtualised with low load/requirements -manual dialing/~5 agents.)
Hi! I noticed that you were listed as the winning bidder for this project: https://www.freelancer.com/projects/linux/basic-server-side-kamailio-sip/?ngsw-bypass=&w=f Did you ever get that done? I'd be willing to give you $100 for the same script if you already have it. If you didn't wind up doing it, we can discuss what it would cost for you to do the same project for me. Thanks, Dan May
...Videobridge to provide high quality, scalable video conferences Jitsi Videobridge (jvb) - WebRTC compatible server designed to route video streams amongst participants in a conference Jitsi Conference Focus (jicofo) - server-side focus component used in Jitsi Meet conferences that manages media sessions between each of the participants and the videobridge Jitsi Gateway to SIP (jigasi) - server-side application that allows regular SIP clients to join Jitsi Meet conferences Jibri - set of tools for recording and/or streaming a Jitsi Meet conference that works by launching a Chrome instance rendered in a virtual framebuffer and capturing and encoding the output with ffmpeg We need this to be installed, configured, and handed over to our team Please share previous successfu...
We are porting our existing SIP applications from using the Radvision SIP stack to using PJ SIP. Because of this we are looking for an experienced C++ developers that has worked with PJ SIP. The company is based in Phoenix, AZ, U.S.A so we would prefer a developer that is within the U.S. Any applicant must have at least 5 years using C++. The applicant must know SIP and work in-depth with PJ SIP. Phoenixsoft develops a soft-switch that provides, prepaid, Class4, and Class5 services. There are three application that make up with SIP interface. These are our SIP Proxy, SIP Registra, and the SIP Call Processing layers. All the application run on RHEL/CentOS 7 and are multi-threaded. It is estimated thi...
android and IOS app, the user login using a user and password, sip server has to be hardcoded in the app, once he login a "call Us" button should be there when he presses it just call a hardcoded number on that button, no need for the key bad on the main screen only after the call initiated the key bad is needed except log out, our logo will be provided below is a picture for what am thinking about. the APP's should be uploaded to my IOS and android store.
