Softclient sip työt

I need a new website. Looking for a site based on wordpress that looks like https://www.foragerproject.com. Site is for a startup beverage/ organic food company. All content and some pictures ready. We will also be using some stock photos. Site must be developed on developers server and transferred to our server once complete, and winner is picked.. Content to pick from…....

mini POC WEBRTC-to-SIP architecture backend buildout for SIP server to webrtc (kamailio or janus) 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 [kirjaudu nähdäksesi URL:n]

looking for developer in WEBRTC expert. Allowing one device(the "master") to broadcast live audio to multiple "peer" devices over local WiFi. As the app should work over WiFi LAN only (with no internet access), the signalling sever "SIP" SERVER. Thanks.

looking for developer in WEBRTC expert. Allowing one device(the "master") to broadcast live audio to multiple "peer" devices over local WiFi. As the app should work over WiFi LAN only (with no internet access), the signalling sever "SIP" SERVER. Thanks.

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

We are 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 &quo...

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. )

trying to be able to use SIP native for android . but also the same UI/layout interface if on the PWA it will work the same but webRTC not sure what libraries to code into. device Mac based ID sends to API built with ionic framework/ angular. not sure exacty what to use [kirjaudu nähdäksesi URL:n] jsSip/ kamailio or what not.

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

Necesitamos integrar un webphone a nuestro CRM el cual esta programado en PHP7, los requerimientos serian los siguientes. 1-Posibilidad de loguearse en conjunto con el usuario del CRM 2-Posibilidad de conectarse a central telefonica por el protocolo de comunicaciones sip 3-Posibilidad de realizar Click to call 4-Integración al CRM para realizar y recibir llamadas

9 Orphans Distillery is a small family-owned South African distillery creating unique handcrafted spirits with lots of love. True to the unique adventures that Africa has to offer. The name of our distillery is dedicated to our late father who had this outrageous dream many years ago that one day our family will own its very own winery. The name of the first wine produced at the winery was called,...

I have a voip project need use the LinphoneSDK + Flexisip server. I have setup the Flexisip server but I found if we use [kirjaudu nähdäksesi URL:n] 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 mi...

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.

An application is required where mainly SIP calls can be made. not sure if we need a server in the middle to handle SiP register so webpage onload (profile) It also requires that the development platform be hybridized, specifically in IONIC./GCP

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 VPN from Remote Side via Public Ip and Port. 5) Assign a Local Ip based on the VPN Traffic . from Remote to Cisco 819 Mod...

Signalwire's SIP trunk on asterisk

Se requiere la configuración de una troncal SIP del proveedor Claro en una central IP con la distribución de Issabel PBX. Actualmente, el problema principal es que la hacer llamadas externas no hay audio en ambos extremos y con las llamadas entrantes no se puede establecer comunicación.

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...

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 ...

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...

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...

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...

Freepbx needs to be installed and configured in our premises Server has two nic Eth0 for management which will be public ip Eth1 for sip server connectivity which has local address

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

Добрый вечер. Мне нужно настроить VOIP SIP 3CX. Как вас найти в скайп или ватсапп для подробного разъяснение? Sipgate+3cx

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

ok, we have a Fortigate 60d router. Everything seems to be working great.... except... SIP forwarding (VOIP). We seem to have an issue with External Polycom phones not registering. Also and CHAN_SIP 5060 lines have no audio. Incoming and outgoing calls work great except on external extension. The phone connects to server, but does not register. Always shows unreachable. Need someone with Fortigat...

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) N...

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 fo...

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]:...

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 ps...

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

Web UI in php to create sip accounts in either chan_sip or chan_pjsip in asterisk realtime database

[kirjaudu nähdäksesi URL:n] 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 avata...

[kirjaudu nähdäksesi URL:n] 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 avata...

Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine [kirjaudu nähdäksesi URL:n] and blogtalkradio.com. and [kirjaudu nähdäksesi URL:n] and [kirjaudu nähdäksesi URL:n] with video conferencing part of [kirjaudu näh...

Hi I would like a module that can provision DIDWW products (virtual phone numbers) in WHMCS so I can manage the billing. If required I can provide access to a development WHMCS install, however would prefer if developer had own install. The DIDWW API uses SOAP for requests. Information on the DIDWW API is here: [kirjaudu nähdäksesi URL:n] The Module need to be able to: == Front end ==...

The project is to create SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and fusionpbx. Need a SIP & Jitsi expert to: 1) create a sip/h323 connector for jitsi, so, it can be given to hosts. 2) setting of sip trunk between jitsi and fusionpbx, so the participants from pstn can connect with jitsi instance deployed in our cloud.

I need someone to set up an authenticated SIP trunk on an audio codes SBC that is then connected to a Teams server. I am looking for someone who has already done this before. Not someone that just has experience in these technologies and wants to attempt it.

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 firmw...

What I need is a logo with a stylized script font similar to the reference photo attached. 3 keys things I love about this reference:  1. The font  2. The thick drop shadow  3. How condense it is.  okay, so let’s get into what exactly do I want.  I would like to have the same thing except it’s says, “Get your sip together.”  I also would ...

Upgrade of existing ANDROID communications APP built with Linphone SIP client with new features such as addition of audio conferencing, auto recharge, addition of subscription etc (and upgraded UI). We are currently using asterisk for the backend. Developer has to be good with UI and Linphone and Asterisk. Also creation/customization of new IOS version with the same functionalities as the androi...

I need an Android app. I would like it designed and built. a One way calling app from users to studio connectivity . with unlimited call waiting function

Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support any client Asterisk instances that we man...

We want your services in connecting / integrating Matrix IP-PBX System to our CRM. The requirements are in yellow, I have added other relevant information too. We can take required CTI licence from matrix once we talk on your proposal. Current Matrix System Screenshot attached. I have mentioned my Telephony setup and wish list. The idea of this has arise, as we are developing a CRM. Current Sit...