Application android sip voip jobs
I need a logo for my new VOIP Company. I want it to look cool and modern but I dont want it to look stupid. I like [login to view URL] [login to view URL] [login to view URL] [login to view URL] [login to view URL]
Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for
SIP Client Application for Both IOS and Android prefer to go with Linphone Features : - Audio & HD video calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Instant Messaging with message delivery status (IMDN) Pictures and files sharing Contact list Call History Display of advanced call
...The features/ pages would include 15-30 Pages around. NAV of MF linking for fetching Price (NAV) info (we already have the membership of BSE). Retirement Planning Calculator SIP Calculator other calculators with various drop down Home page: Scroller images Corner: Login call us icon for: request a call back with preferred time slot & day 1. About us:
This project on creating Tools and Calculators for an Indian financial website. Examples of tools will be of Home Loan calculator, SIP calculator etc Please share relevant work references
...com/OpenNMS/opennms) is an open source monitoring and systems management platform written in Java. It includes support to monitor SIP devices registered to and asterisk server and it does it by connecting to Asterisk's AMI interface and querying for SIP registrations. If the list of registrations change, OpenNMS generates an event or an alarm. OpenNMS depends on a
Good day, I am looking for a resource to configure queue metrics and freepbx, this is a new clean setup, new boxes. This you will start with clean setup. We have already a sip trunk provider and there accounts need to be setup also Both boxes is already loaded, we only nee to finish off the setup and configuration. We wish to extent both products
If possible, I would like to route an inbound route based on TRUNK and NOT based on DID. FYI: My SIP provoider is NOT sending the DID, so the routing should go based on TRUNK (thus when an incomming call on trunk A comes, then it should be routed to extension 200, when an incomming call is comming on trunk B, it should be routed to extension 201 ect
We are a hosted cloud voip/phone system provider and are looking for a 2 page voip ad brochure. We currently have a design that we would like to emulate and just need the design created in either, adobe illustrator, ms publisher. We will provide ALL wording and design layout.
I need physical Turkish phone numbers for Microsoft Azure verifications. I can buy up to 20 phones per month but all of them must be non-voip and didn't use on Azure before.
Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.
We are looking for a VOIP expert to help us to figure out away to make unlimited phone calls to the United States and Canada for free. With NO volume restriction. We know its possible if you know of a way. We would like to hire you as a consulting. The quality and reliability must be high. If you know how to do a proxy server over google voice, skype
Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.
Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing
Hey Need to add a new feature ASTPP VOIP BILLING ASTPP use freeswitch pbx, but he generate diaplan. wee need add function pitch shift
Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing
Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing
...someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones like cisco, Unifi VoIP phone and Polycom. Thanks
Hello, We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough
NAME IS NEEDED. NOT A LOGO!!! I NEED A NAME SUGGESTION Study Cafe Name needed! - This is a space where you can study and get sip of coffee and some desserts. - space is mainly for studying (and maybe working) - there are space for them to do team work - there are space for customer to rest too - please see some example pictures.
I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.
...an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5. - The SIP signalling shall transport a 180 RINGING message without SDP parameters meaning the originator should
...API 2. Call is initiated via Twilio Outbound Voice API using a Phone Number. 3. When call is answered, the call is forwarded to another Twilio phone number that uses Elastic SIP Trunk. It is attached to our IBM Voice Agent Service. 4.. Once Voice Agent is connected the user will speak to it., then be transferred to another phone number. We are looking
I need a developer who can develop VOIP apps for ios and android. App features, look & feel and functionalities will be very similar to rebtel apps. [login to view URL] I will provide design files.
I wanna run voice campaign so i need a solution from where i am able to run voice campaign.
We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.
Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing
We are a company specialized in termination. We have many destinations with several operators to handle , each has his own unique system. To keep track of our business and to be able to make sure everything is working properly, we need an expert in cloud configuration to manage the cloud configuration. We know exactly what settings we want to apply, all you have to do is the proper configurations....
I am a asterisk Expert who can easily understand about auto dialers, i want to make a web application that allows autodialing using voip
hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [login to view URL] if you can make it please bid
Need to develop SBO VPN to compress VoIP bandwith,
Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5
I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.
...to send call server to local voip gateway like server in USA VoIP gateway in Bangladesh server to Bangladesh Openwrt Router connected BY VPN and VoIP gateway connected to openwrt router so i send call from server its work fine but i need some one who can help me to create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want
if you developed any Android Voip app and have source code, please bid. I will discuss about it in details and let you know what i need it. Thanks.
We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand
...[login to view URL] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order
if you have Android Voip app source code, please bid. I will discuss details . Thanks.
I need to add call history and contacts function.
I need 10 minutes help (remote assistance /Teamviewer) to resolve an issue with Miktotik Queue with mangle for VOIP
I need help to configure Mikrotik queue for voip traffic sharing my desktop in teamviewer.
Application key features: * high definition audio and video calls * audio conference calls with various participants * pictures and file sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers
I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP
We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your
Hi There, Looking someone in order setup A2billing + Callshop module, calling card, and ivr setup. Thanks to reply if you got some experience on it
I need for someone to build a simple 3 page wordpress site and for this person to have completed numerous twilio api jobs.
I need someone available to build ios app. This app will have chatting(basic feature: text/video in group), VoIP, whiteboard, payment api integration. Please don't hesitate to share your skills and experiences.
Hi We are looking for freelancer who is aware of complete integration and workflow of GrandStream HT813 with Issabel. Issues to resolve Call Progress Dial Tones to be set for Airtel (India) Call disconnection (FXO Termination) to be set Other settings Secure Connection between device and server Complete Trunk related settings in Grand Stream and Issabel (Elastix) Only bid if you have complete ...
...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m
