Blackberry sip softphoneKazi
Necesito un scriptlet que me permita llamar a un número telefónico, ponerlo en espera, llamar a otro número telefónico, unir ambas llamadas para iniciar una conferencia tripartita y después dejar en llamada a ellos, cuento con un servidor FreePBX y debe ser web, por lo tanto he buscado hacerlo con sipml5, y jssip, pero en todos solo he logrado funcionalidades parciales
...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/Whatsapp calls. Functional flow 1) Calls originating will send to Whatsapp gateway 2) Whatsapp gatewa...
Are you able to configure Zoho Desk with WebRTC clients to use an arbitrary SIP provider? You are the person for the job. You configure Zoho Desk Phone integration with arbitrary SIP provider we are using currently. The job is complete if we can make outbound calls using the Zoho Desk WebRTC client and receive inbound calls from SIP provider in Zoho Desk and process them in a example call flow and answer by ZohoDesk WebRTC client. We are currently using Twilio phone integration but we believe at time of this post Zoho does not have a way to control what trunk is used in Twilio.
We are looking for an in-house call centre solution. We will provide our own server. The objective is to have our own PBX system as well as call centre system for the customer care department. The tasks include: 1. Install GoAutoDial or Issabel in the server. 2. Connect with the telco provider (SIP link) and configure hotlines numbers for both call centre and PBX. 3. Configure IVR, ACD as per the attached IVR flow. 4. Configure voice recording for all call centre agents. 5. Configure the IP PBX as per the attached requirements. 6. Create the API so we can use the feature of "Click to Dial" from our CRM (web based application) 7. Info-popup, that will fetch the customer name from the CRM DB when a call comes in. This will allow our agents to greet the caller by name....
We are an android developer We have an android app that receives video calls from hardware ( intercom system). Our app uses linphone client to receive calls. - Linphone Android works perfectly over Wifi network, but when we switch to 3G/4G it stop working. No SIP registration happen or any incoming call. - Switching of network needs to work smoothly. Need to solve this problem and only expert who already solve this problem place the bid. We are an Android developer and we are looking for quick solution. DO NOT BID you're not expert on Linphone Android. Our team will assist you with anything you might need.
I need a programer to change the layout of a open source SIP VoIP app and publishing it to the Google play store an apple iTunes.
Hello there, I want a Sip Server or Sip Phone ready made code. Functionalities that I need the most is: 1) When one user calls another user through our server and that user does not exists server must call one exe. 2) Sip phone must have a feature to accept multiple calls on one device redirecting calls to specified list of microphones and speakers in a text file. Note : We do not want to develop it from scratch. We do not mind any technology working on this but there are two terms:- 1) Code should give us a good control and should have good code standard. 2) Code should allow us to run server and sip phone on Windows system.
I need a softphone with my brand and excellent sound quality with payments options and security
I need android walkie talkie app with GPS tracking. Backend with SIP protocol, and frontend in Laravel for manage user/channel/maps/video.
I am looking 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 the bullshit. I will provide sample layout of homepage and dashboard to build
I'm looking for somebody that can install siptrunk on my aws. The problem is that I need a static ip but i can not fixed it.
Hello, I need a Porte's value chain analysis on Blackberry (one page only) no more than 400 words, for tomorrow night.
We have a new Cisco UC560 that needs configuration. Device - Cisco UC 560 Needs to configure 1 SIP trunk and 6 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 Extra $$$ for quick completion
...caller details on your soft phone or desk phone while making calls. * Incoming and outgoing call events automatically logged inside Zoho Desk. For local soft phone we are currently using Bria 5 by Counterpath. We are also considering Zoiper 5 PRO. As supplier of managed voice services, we supply our own SIP trunks / hosted extensions that integrate perfectly with both platforms. We look forward to integrating with Zoho PhoneBridge in order to automate call logging process within Zoho CRM. Explanatory documents relating to scope of work attached. Please contact me should you be able to assist....
Hi Freelancers, We want to a Windows 10 Based SIP and XMPP client developed. You Will design and develop the software. Yes need to design a logo and icon for the product. The branding and licensing will be fully owned by ALSOFT. I will give the name later. We have two software. One is open source Linphone SIP client. I don't like the design of Linphone software. I like the Bria Professional SIP phone interface. One the other hand we use spark as a XMPP client. Spark is really a powerful XMPP client. You can use modify Spark and add the SIP functionality there. Basically I need merged version of Linphone and Spark. We use FreePBX as SIP server and Openfire as XMPP server. Regards, ALSOFT
Hello! Looking for someone who can construct for me a SIP connection system to make VOIP calls to west Europe (spain, france), Turkey and Palestine (most of the calls will be to turkey). Im not looking for something ‘large scale’, just to make a few outbound calls, with the ability to customize my CallerID freely. I have little knowledge in the technical aspect nor the servers to host it, so everything needs to be provided by your side. Contact me if you see yourself viable Oliver
Hello! Looking for someone who can construct for me a SIP connection system to make VOIP calls to west Europe (spain, france), Turkey and Palestine (most of the calls will be to turkey). Im not looking for something ‘large scale’, just to make a few outbound calls, with the ability to customize my CallerID freely. I have little knowledge in the technical aspect nor the servers to host it, so everything needs to be provided by your side. Contact me if you see yourself viable Oliver
We are implementing Jitsi sip video gateway using Jibri and Pjsip or Pjsua We need help troubleshooting: While initiating the call, Jibri is accepting the request but Pjsua/Pjsip is not starting. Need help to troubleshoot the same. We can provide error logs and necessary access.
I need you to develop some software for me. I would like this software to be developed for Windows using C or C++. The sip stack application is base on PJSIP SIP. standard
I have a business that needs a mobile. I'd like to have a mobile application done for my business to be used on iOS, Blackberry, and Android devices... Please help!
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[�...
Make a mind of a travel website with following details Users may browse the website as guests or as registered users but only registered users can create bookings. • Users can book f...arrival airport o Hotels o Duration and days of travel • The system will allow the user to sort the returned information based on price, and direct / connecting flights • The system allows a maximum of 5000 users to be on the system simultaneously • The system needs to be functional on desktop browsers such as Chrome, IE, Firefox and Safari as well as mobile devices (iPhone, Samsung, Huawei and BlackBerry) for the same list of browsers. • The system also needs to maintain security by ensuring that personal and/or any financial information from the user is not leaked or comprom...
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
