Web java based sip client dobble jobs
I need you to de...functionality. Would need to be able to host up to 35 users on a single meeting and many meetings concurrently. I am open to the use of function Technology like web RTC or turn servers/sip. Please provide your recommended architecture and examples of your past video conferencing platform development in your response to this post.
I need someone to configure a SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working a...SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working and I have other devices connecting remotely. If you can config the sip settings that is an added bonus.
...wholesale clients, - Setup and updates rates - Configure Plan Packs - Termination provider configuration - Setup for SMS - Configure our dialing plan to route calls to your SIP gateway: - Create wholesale clients, from whom you will be receiving calls: - Setup IP PBX management - Migrate clients from current setup a2billing/Freepbx - Secure windows
...I need a FreeSWITCH (FS) configuration for the following functionality: 1. Interconnect internal endpoints 2. Link with an existing Asterisk PBX 3. Link with an external SIP trunk for incoming and outgoing calls 4. Configure basic CDR, voicemail, IVR and conference 5. Configure presence/BNL 6. Configure chat on Linphone endpoints (optional)
Need a developer that has a PPT (Walkie Talkie ) application that we can embbed into other appllications. The PTT platform must work with a Sip Server like OpenSip or Asterisk. Users can connect to server via any Mobile Internet, Wifi Internet or via Bluetooth hotspots that will be setup for users connection. This project is not a Design from scratch
...post. I have a dedicated Linux box with Ubuntu [url removed, login to view] installed on it. Have also installed Asterisk but have not configured it. I would be looking for someone to setup the SIP information and install a soft phone and get it to dial out. Furthermore, to protect my PBX server I would want whatever possible basic safeguards to be in place (ie. changing
...products involved are needed. Inbound calls will be delivered using VOIP so a computer with a SIP client such as X-Lite or a mobile phone with a SIP client such as SoftPhone will be required. If you do not have previous knowledge of working with SIP phones we can assist you in every way. A reasonably fast internet connection is also required to
I have a custom andriod voip client done using linphone. the job was not 100% completed. I need to fix the logo and turn off settings , turn off the sip address when dialing a number, and make the app recognize only the last 5 digits of the phone number. I only have the app no source code.
...runs on gnu/linux as a service and has some hardware button handlers for PTT,change room,mute. Software must be P2P serverless or use SIP or mumble voip [url removed, login to view] least 9 user connection supported. We need just SIP soft phone software which has these features; - on/off switch for PTT mode and VOX mode working - PTT button for if PTT mode is on
I need an Android app. I would like it designed and built. To combine linphone(sip) and conversation xmpp (messaging/chat) into one app. Both app have source code at GIT . linphone by Belledone and Conversation by SIACS.
Project description: We need FreePBX or any powerful PBX phone system what is very stabl...time ONLY those phone number to analyses if the number exist or not. Must be absolutely NO CHARGE making by each call and write back into database FAIL / exist number! The sip trunk have be minimum 250 channels that the dialer can call per second 250 numbers.
- Asterisk Setup and config as per requirement. - SIP server for Voice, IMs, Video calls with codec setting, disable other things. - Config to use FULL server resources ( Ram config/ Network config ) - Load balance high volume calls. ( currently we will use only 1 server, but needs to be modular for load balancing ) - Dial plan configuration
I need an Android app. I would like it designed and built. need to built android app client intergration sip & xmpp
We essentially would like a web based interface to input configuration data and it outputs pre-formatted XML files for VOIP Phones (Yealink, SNOM, etc). They can then connect to this web server and collect their provisioning files that have been generated and it configures them based on the output XML configuration files this will generate. Initially
Hello We are looking for a software developer to modify a current and supported open source VOIP software capable of making sip calls. The choice of software would be up to you, however the software MUST -Be compatible with IOS and android (newest versions). -Allow the user to select whether to use the integrated caller or use the VOIP software after
...android app - SIP client for voice, ONVIF protocal for video - delivery in 7 days. Developer must have skill for SIP and ONVIF. If not so, don't bid on here, please. I have screens already, so I will provide through chat. Again, don't bid man that have not skill for SIP and ONVIF. On the top of your bid string, write "SIP & O...
Installation of GoIP SMS Manage Server on CentOS linux (Elastix 4). I alreary use an Elastix Predictive Dialer whith a GOIP8 as a GSM SIP Trunk. I whant to send an SMS message automatically through GOIP when the campagn ends to retry 2 times each celphone number without success.
...app - SIP client for voice, ONVIF protocal for video - My budget is 300USD and delivery in 10 days. Developer must have skill for SIP and ONVIF. If not so, don't bid on here, please. I have screens already, so I will provide through chat. Again, don't bid man that have not skill for SIP and ONVIF. On the top of your bid string, write "SIP &...
...should ring simultaneously on incoming call) * 1 external SIP trunk provider for incoming and outgoing calls For outgoing calls, all numbers dialed on internal endpoints that match full international number format (e.g. +447712345678 or 00447712345678) should go via the external SIP trunk, all numbers that are longer than 3 digits (i.e. that are