Solaris sippekerjaan

I need to turn an Android cellphone as SIP to GSM Bridge of Gateway. Traffic from SIP must be translated to GSM channel. You are free to choose how to implement this feature. All that is required is that in the end the Android phone will be acting as a SIP to GSM channel/trunk/GW and other SIP devices can connect and make phone calls using this. From my side i can provide you with Linux firmware and documentation for few native SIP > GSM GWs. This is the first thing which is need to be developed. Depends on result of a job contract can be signed to future cooperation.

I have an asterisk server which has oem set functionality which I want to be accessible via an android app. Below are the requirements. Allow users to be able to call each other within the netowrk via UserID - users must be able to call each other at anytime i.e. constantly connected to the SIP Include speech synthesis in the app - so user can type in what they want said at the start of the call and it gets sent to server Ability to set Caller ID internationally View call history Set voice pitch dynamically through app if possible Receive DTMF dial tones as sent to server by callee Must be modern design and run smoothly. Must include FAQ with details on how to use the app Must include help page with contact information On startup: must include a progress bar with audio file play...

Hi Eremin P., I noticed your profile and would like to offer you my project. We can discuss any details over chat. We need to make Kamailio to work with MS Teams and a SIP provider. Part of the configuration is already done, need to finish the number rewriting, RTP proxy and routing part. (Teams is already connected)

Hi Solaris Technologies, I noticed your profile and would like to offer you my project. We can discuss any details over chat.

Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.

Hello Folks, I am looking for Linphone expert. We are done with entire customisation but we are facing issue in call. Below is exact issue Apple requires the use of PUSHKIT to receive silent notifications when the app falls into the backgr...am looking for Linphone expert. We are done with entire customisation but we are facing issue in call. Below is exact issue Apple requires the use of PUSHKIT to receive silent notifications when the app falls into the background and the app is sleeping. PUSHKIT requires an IMMEDIATE call to CALLKIT which starts ringing the call. The call is now ringing without the SIP invite having arrived at the app. The app registers to the SIP proxy after calling CALLKIT. The INVITE is send AFTER CALLKIT starts ringing the app. Please bid if you ca...

Title: Sip or Spill - A Bachelorette Party Drinking Game NO AUTHOR Dimensions: 6 x 9 inches Needed Final Formats: PDF, PNG, JPG (RGB & at least 300 DPI), and Ai/Vector + Layers/Fonts (You can zip the final files). For the title, I would like the font to be bold, but girly. I have attached two sketches of some ideas I had for the cover, but I would like to see what you come up with yourself. The font should be clear and easy to read with no shadowing affects. The text should be clearly contrasted against its background. Please use cohesive colors that blend well, it should be eye-catching and vibrant but not distracting or too busy. I want the big illustrations and the title to be the main focus, the background should not be super busy. Do not make the background white. The ill...

Hi Solaris Technologies, I noticed your profile and would like to offer you my project. We can discuss any details over chat.

Ciao Raffaele, ho visto il tuo profilo e da quanto letto suppongo ci siano i presupposti per sottoporti un lavoro, anche con la semplificazione della lingua, che dovrebbe essere per noi un concept da discutere col cliente per capire se renderlo produttivo o meno entro 1 mese. Le tecnologie che verranno toccate saranno principalmente webrtc, js e sip.

my project is simple, Install and configure stable RestComm smsc gateway ( ), so smpp to sip message should works with good performance. Centos os is preferred but you can work any os.

Hi Solaris Technologies, I noticed your profile and would like to offer you my project. We can discuss any details over chat.

...want a website where people can register with the following: Name City Mobile Number Email PAN Number Date of Birth Once registered, they can log in and be able to see: a. List of all Mutual Funds Units he is holding in India b. Current Unit Value of the Funds he is holding c. The net worth of his Mutual Fund portfolio d. P/L for each holding Further, he can filter the above info on "All" or "SIP" or "SWP" or "Lumpsum" etc. with respect to payment mode. We do NOT need any units purchase feature or reedeming of units. Just viewing the details. Clearly mention the following in your bid: 1. What technology would you use? 2. Which API would you use? 3. What would be the cost of the API? 4. What would be your development cost? 5. Exac...

We need drawings made for a 3 bedroom 2 bath SIP home.

-Install FreePBX on a LAMP server and configure trunks.

i want to make a SIP dialer dialer have to work all network codec must be g729

Set up Kamailio in front of asterisk server to proxy webrtc/tls /tcp / udp to asterisk server on private network with failover. - rate limiting sip traffic by source IP - dropping malicious / invalid packets - integrated with APIBAN () - MySQL integration to be able to adjust config (like primary / secondary / tertiary failover etc) - auto install script so this can be reinstalled / moved / etc easily with minimum effort. - Bonus for early start / completion within 24 hours

Hi Solaris Technologies, I noticed your profile and would like to offer you my project. We can discuss any details over chat.

We need a phone vendor for a SIP migration. Customer has existing POTS lines and they are migrating to ATT IP Flex connection.

1. Linphone registration a) Linphone sends REGISTER message containing its PUSH token via Kamailio to the PBX b) When Kamailio receives the OK from the PBX regarding the successful registration it stores the Linphone push token and username Incoming INVITE to Linphone PBX---->Kamailio a) Kamai...INVITE from the PBX and does not forward it to Linphone b) Kamailio sends instead a PUSH to Linphone to FCM (google firebase) c) When Linphone receives the PUSH it issues a REGISTER message d) Kamailio catches the current contact address from the REGISTER message e Kamailio now forwards the INVITE from step b) to the current contact address of Linphone f) The usual INVITE from step a) proceeds SIP-INVITE ----->[PBX]----->[Kamailio]push------>(android[UA/SIP_REG-PBX]-...

Looking to build a setup where. 1.) UA /WEBRTC client registers to kamailio 2.) Kamailio is registering directly to PBX for all SIP. extensions. 3.) setup where Kamailio doesn't use RTP engine (not sure if i can) need to test config is almost there Need some assistance/tweaking. as i dont know.

Page fixes milestone for Total Indie platform

hi, We need a App that can make and receive calls through sip client. development should be in xamarin. thank you

Seeking for someone who can help with a sip integration. Please only people with experience in VoIP and / or jitsi. Thanks

my cafe name is "Eat 'N' Sip".intrested freelancer can bid on it.

We're looking for a Shopify Design expert to design our new Shopify store for a Paint and Sip studio. This will also include GuestManager Shopify plug-in as the Event Ticketing platform, which will also need designing (eg. Ticket design). We are looking for someone with extensive graphic design experience that can also work with the Shopify theme to produce the live site.

Hi there I have a Softswitch name ASTPP V3.6 which is running on freeswitch There is failover gateway options which I try to use it but its not working want solve that. And also one more issue that sometime SIP account in unregistered automatically. Thank you

Our Kamailio server configuration is not working with sip registration. Need reinstall of the Kamailio server. Please apply only if you have already worked on Kamailio

Hi As per description, I need 1x UI Back end (Dispatch ) with our logo hosted on our server and domain 2 x Androids Mobile Apps (1 for the Driver and 1 for the Passenger) 2 x ios Mobile Apps (1 for the Driver and 1 for the Passenger) The Companies Colors are RED, White and Black THIS IS OUR CURRENT APP WANT SOMETHING WITH SIP OPTION ADDED FOR A SOS BUTTON FOR EMERGENCIES AND PAYFAST PAYMENT GATEWAY.

Webphone for integrate to our webbase sales management system. Functions : Answer, Hold, Transfer, Reject, Mute API Strong Installation Documantation (Word, Video)

I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory

...Phone. The target uses ARM architecture, runs Pine64, an embedded platform and has Node.js based API running on Linux. Deliverables for the POC: We would like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsite APP communications. Establish a webRTC to SIP(PSTN) connection having two-way audio call and local VoIP connections (Guard Phone). If an alternate viable method is proposed we can consider it...

Need help integrating DR3000 repeater with SiP based system

Hacer una práctica en Linux Debian en su versión 10.1.0, Asterisk 11.25.1, softphone Zoiper o Blink en sus distribuciones Free y configurar la Seguridad en SIP dentro de un mismo escenario (Máquina Virtual), Wireshark, VirtualBox

hello Desginer We'd like to design 2 banners for Collagen Sip Straw with following contents ( key message) 1. Radiant Beauty with Collagen Sip Straw from Japan for 4 weeks 2. First Collagen Straw in The world Reference: - - Thank you

Looking a freelancer who has experience in building International SIP ConnectionSIP system

I need to run the my voip server with Asterisk. I also want to run the x-lite softphone from the VPS as well. What I need: Set up Asterisk on my server. Configure my SIP account from my number provider to voip vps. Configure

We are looking for a skilled developer or team to develop a simple one click remote support for our team and custom...open source solutions or from scratch according to your preference as long as it includes the expected , I wish to mandate you to take care of my private server hosted at OVH in public cloud, I set up and installed the solution "GoAutodial" in v4, unfortunately I meet bugs with "Kamailio" on the telephony part in webrtc in the browser. I explain myself, everything works correctly, the SIP is well connected in asterisk and I can make automatic calls on the campaigns, unfortunately one call on two I do not hear sound for the interlocutor. I tried everything to debug it, I can't manage it. I don't think it's a big deal, I'll let y...

I have Jitsi configured on my server but facing some issues with installing Jigasi for Transcription and SIP. As of now, I need to enable Jigasi for Telugu language, so need to understand how we can configure multiple languages in Jigasi. Google speech to text account is free for 1 hr so, you should be able to test and configure using that.

Configuratiion Push Server to wake up our mobile SIP dialer to a pBX

press prefix 9 or 90 it should take sip line of yeastar S300 PBX and call for Outgoing

Hi Solaris Technologies, I noticed your profile, I would like to discuss my project with you to see if we can work together. Thanks!

For a voip and sms provider based in Europe, we're looking to generate sales based on SIP-trunking of SMPP trunks for SMS. Connect with the VOIP providers of your country and introduce our company. Generate the contact and introduce ourselves.

...router/firewall donde quiero que funcione BigBlueButton. Como el operador de comunicaciones (O2) no me da IP fija accedo al exterior utilizando DDNS. Todo esto ya lo he ehcho y en principio funciona pero tengo un problema: Los usuarios externos se conectan pero la prueba de audio no funciona. He oído que es un problema de que los paquetes UDP no cruzan el servidor aunque he visto que el mensaje SIP que devuelve mi servidor va con la dirección IP local y creo que e spor eso por lo que no funciona. ...

...ideal for a morning water workout, to work on your tan while you sip your favourite beverage, or just people watch. As day turns to night, relax on the comfortable deck loungers with spectacular views of the city. Sip on a drink of your choice from our large selection of premium wines, beers, signature cocktails, join us poolside for the daily sun-downer cocktail ritual. - We call it "The Edge" because it was decorated with a transparent glass perched on the very edge of its own infinity pool which makes you feel like you swim to the horizon. USP 1. A tropical bar style & a transparent glass on the edge create a unique experience to the guests. 2. Variety of food & beverage selections. It’s the place you can have a sip on a drink of your choi...

Building a custom android app via ionic. adding linphone SDK in capacitor . UI layout is done just need to build out the core classes and connect the dots. and do SIP testing SIP testing Make sure SIP Call works 2 way audio Make sure SIP DTMF works (SIP-INFO) Make sure RTP is flowing correctly Make sure SIP Inbound CALL is working [do we need a push notification server] Make sure SIP Register is working with the correct Timers in place .

Requirements : -Build a VOIP portal with concurrent calling for use with SIP trunk providers.

I want to make my PBX outbound SIP calls to pick the next available SIM slot in cycling manner if the previous is busy, Setup SMS sending & receiving using SMS API, Call Credit Limit management, SMPP for SMS. Experienced freelancer with dinstar uc2000-vg can bid

Hello, Freelancers I want to deploy Android and iOS app using React Native. You must have experience with SIP JS. Please share your app using Thanks

What we need from you? Need a story or concept to promote the core idea mentioned below that can be expressed through Sand Art movie. Approximately 1000 to 1500 words to explain the idea. No google concepts or Youtube Plagiarism Only original ideas welcome. Core idea: We need to c...promote the core idea mentioned below that can be expressed through Sand Art movie. Approximately 1000 to 1500 words to explain the idea. No google concepts or Youtube Plagiarism Only original ideas welcome. Core idea: We need to communicate the concept of Mutual Fund to general audience in an innovative and different format than general Mutual Fund companies. You can use the below features: SIP Tax Savings Discipline and Patience Short term and Long term investments You can submit in PDF or Wo...

Hello, Freelancers I want to deploy Android and iOS app using React Native. You must have experience with SIP JS. Please share your app using Thanks