Sip softphone j2me jobber

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 like to have a muted emphasis on “SIP”, (kind of like the “YOUR” in the reference pic)but open to playing around with the  placement. Do you, but also keep that in mind.  This logo is for a cocktail line, hence the word play. Which is also why the logo needs to be playful, simple and full of personality.

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 android application (above) using Linhphone and same APIs as the android version All APIs and wire frames would be provided for the UI.

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

...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 manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we are more secured against breakdowns, thus we want to have a redundant setup with a failover IP in the future. It is also very important that our systems do not have lag or quality challenges, neither for our agents nor...

...Plan) – 200 nos DID, 0612-6604400 to 6604599, 30 Channels 2. Using Matrix EPBX (IP-PBX) – Screenshot attached. 3. Using SIP Phones – Alcatel-Lucent 8001G Open SIP Phones. 4. Using prop soft phones of Matrix (Varta) – Inside and outside office – Have static IP in office to work on port forwarding 5. PRI is parallel looped to XTEND VLS recorder – which in turns records all calls in the system hosted in our office. Having call details and recording there is important, as it reconciles orders our agent puts. 6. We use DID logic, so all people in office have direct numbers. 7. EPBX gives us parallel ringing, so one extension can be activated at three places (Mobile Soft phone / SIP Phones / Computer Soft phone etc) So, we want our PB...

Hi Friend, I need the profesional for helping me setup system: - Kamailio load ballancing : Sip/WebRTC - PBX use Freeswitch core with Postgres Database - Postgresql database is shared on every freeswitch - Inbound call route to kamailio, outbound call operate on every freeswitch node The result of project: - System setup like description above - Can use Softphone use TCP protocol to make/receive call - Can use client use WebRTC to make/receive call - Help sizing how many call server can serve ( with hardware 4CPU/8G RAM Network 100Mbs) Prefer Team/person who have already did this job in the pass.

We are building a Park Model RV, and need to design the exterior walls ceiling, and floors.

configure asterisk on ubuntu SIP Configuration ( incoming outgoing) IVR Menu (welcome menu, directory menu to (call groups) Create Call groups Call back request in case agents are busy for over 5 minutes) Enable Call transfer call record call waiting call transfer call hold tunes

...marketing software which can be integrated with above software. 8) Sales funnel with landing page where the data is collected in single place. 9) Add financial calculators with infographics. MY CURRENT WEBSITE IS I WANT A MODERN LOOK WEBSITE LIKE For the financial calculator you can see My website needs to be well connected with my software as it is now and I want the builder to keep SEO in mind while re-designing the website. Develop a website for me, with complete website design, development and database. Website Development Platform: Complete website design will be made using HTML & CSS, server side will be made using PHP & for database SQL/MYSQL will be used.

Hi, We use Electron for a desktop telecommunications softphone app and are looking for a developer who knows Electron and the telecom VoIP business. Need to know Javascript, PHP & CSS We have 3 versions PC, Mac & Linux / Chromebook but I can provide a Mac environment if you do not have one. I Will pay by the hour, it will be ongoing work for a long time. Thank you!

Installation of GOautodial on your server With full customizations and configuration of campaigns, scripts, calling hours, carrier/trunks (SIP/IAX), lists/leads, statuses/dispositions, IVR configuration, and other dialer features you need. WebRTC softphone configuration(kamailio). System instalation in VULTR.

...We 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/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gatew...

configure asterisk on ubuntu SIP Configuration ( incoming outgoing) IVR Menu (welcome menu, directory menu to (call groups) Create Call groups Call back request in case agents are busy for over 5 minutes) Enable Call transfer call record call waiting call transfer call hold tunes

Development of a SIP to Whatsapp gateway required. Either standalone system or module for Asterisk/Freeswitch.

Calculate the loads in Fusion 360 or AutoCad or any other CAD program that is suited to this task. A SIP (Structural Insulated Panel) is EPS (Expanded PolyStyrene) or "syrofoam" which is laminated to plywood, or OSB (Oriented StrandBoard) or other materials. The size is 3 inches thick (76mm) (of EPS), + adhesive, + 1/4 inch (5.3mm) "hardboard" x 1220x2440mm

I need a banner image similar to the example attached. Will you please use the colors in the file titled SIP? Will you please place my logo where the logo is on the example? To the left of My Headshot, will you please put my company name "Serve, Impact & Prosper?" To the right of My Headshot, will you please put the following (3) bullets... * Create Systems... * Grow Your Team... * Save Time & Earn More Money... My name is Jay Jackson

Hi Henry T., I noticed your proposal for project https://www.freelancer.com/projects/Linux/SIP-Whatsapp-gateway/details Can you please do the same work for me? I need VoIP Whatsapp gateway module for Asterisk or Freeswitch.

Need to make Cisco504G, Linksys 942, Fanvil X7 Series and Grandstream phones compatible with FusionPBX. Set Up Provisioning to manage… Sip Registration Local Date & Time as per Client’s Location LCD Screen & Screensaver (On/Off/Timer) Idle Key List Missed Call Key List Off Hook Key List Dialing Input Key Progressing Key Connected Key Start Xfer Key Start Conf Key Conferencing Key Releasing Key Hold Key Ringing Key Shared Line Key BLF Key Programmable Keys -Intercom -paging - Call Park - Barge In For all 3 phones, extension unregisters randomly Cisco SPA504G & Linksys 942 Issues Bring Day/Date/Time from Context Attended Transfer doesn’t work Blind Transfer Works Program Intercom Button with *8 Grandstream 2600 Series Ringing Issue, call doesn&rsquo...

Looking for twilio expert who knows how to create sip end points remotely. or knows how to work with twilio programmable voice api's

Hi Swapnil thanks for configuring USA SIP Trunk I may need your help tomorrow when actual operations start.

...and advising on the solution Logging and keeping records of customer queries Analysing call logs so you can spot common trends and underlying problems Updating self-help documents so customers can try to fix problems themselves Testing and fixing faulty features You should have the following technical experience: Linux and being able to install software and Troubleshoot on Linux SIP trace and understanding of SIP protocol Experience in multiple class4 softswitch vendor Experience in SQL and or postgress...

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: h...configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: I may probably also need help with G729 codec configuration

I would like to setup a sip trunk inside twilio for outbound calls only, with a different cheaper provider (zadarma for ex.). If its possible transcribe all incoming and outbound calls to portuguese (brazil) and sent to AmoCRM(OpenAPI). Also i need to setup a IVR that has a greeting welcome message and then if it rings about 8 times and nobody answer, then the ivr takes again and a recording says: All operators are busy right now, please press 1 to receive a Whatsapp Message and continue (if it's on landline, manually insert mobile phone) and send the template automatic or Press 2 to receive a call (then integromat create a task (list) into AmoCRM(openAPI) to a user to call this contacts that we have missed calls.

We have bought sip trunk with FPBX. Now we need to deploy the pbx on a cloud. I understand the configuration but need someone with experience to guide us. The main work is to configure outgoing routes, maybe incoming routes. Once that is done we want the application to communicate with the server for phone calls (this we will do ourselves). I need a dynamic and quick to respond personality who can finish it ASAP.

Hey I have a 17 user environment that I need help setting up. My preference is 3cx and I will be hosting it in google cloud. We have 17 users spread over 2 sites currently, however, we have 32 sip numbers. Each employee has a direct line, then there are 2 receptionist phones (as well a 1 wireless phone for when the receptionist has to leave her desk) and the rest are on standby for expansion. I need assistance in provisioning the desk phones as well as setting up the call routing for the PBX. As far as features are concerned I need most of the Pro features enabled, especially the switchboard feature (receptionist desk settings) I am assuming from the feature list that I need the Pro version but I will be happy if I can get away with the standard features. I have listed t...

Hi all, I am looking for a system admin to install A2Billing on my CentOS 7 server. I have tried installing it using the following guide but it did not work: I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,exten,priority=a2billing-did,s,1 I have tried this installation guide and some others on 2 different servers: one with Asterisk 16 and FreePBX 15, and the other with Asterisk 13 and FreePBX 14. I'm looking for an engineer with A2Billing experience so that to work on this task of installing A2Billing. It will be for the engineer to choose the server to use from the two I have mentioned above

We would like to connect one Microsoft Teams user to a DID on an Asterisk box via SIP

I need to configure my IP phone Cisco CP 9971 to work with SIP provider

Cần 1 người viết hàm lưu ảnh trong game j2me,Nhận hình ảnh từ game rồi lưu lại vào 1 thư mục

Hi developers, We are looking for people who have experiences in the integrating Zendesk with SIP Softphone. We are in the midst of development and would like a hand to help speed up the development. Please refer to the requirement doc attached.

Hello We have an urgent need of a person who can handle Issues / Queries and enhancements in Vicidialer and also able to handle Connectivity with SIP/EPBX/GSM Gateway Connectivity with VICIDialer

Need a windows application that connects to sip server and sends fax.

i am use nutshell crm ,and CounterPath bria5 softphone, have possibly integration two system? Contact Synchronization - Inbound calls from external numbers trigger a contact lookup in CRM, and contact details are added to nutshell Contacts. This way, the caller name is automatically shown in your phone display when you receive the call. Call Pop-ups – the customer record is brought up to you automatically when you receive an inbound call. Call Journals – Calls are logged as call records in the CRM. Create a new contact automatically when a call is received from an unknown number.

Hello, I want to build API [Java or php or C/C++] to call to Asterisk Server by using Android also from Windows [Browser]. The API will be one for both Android as well as Windows browser. I have purchased Asterisk SIP server users [Two number] and they gave me two extensions alongwith Server IP and password. Will shore the details after award the project What I need a API to integrate with Android as well as call from browser. The requirements: 1. Need API to Generate a call and receive a call [Both from Android and Windows Browser]. 2. Call recording 3. call Duration Log 4. Source Code after checking the Demo.

I am looking for Linphone branding, later will do is an open source SIP Phone, available on mobile and desktop environments (iOS, Android, Windows Phone 8, Linux, Windows Desktop)

...such as houses. The walls of a Panelok building are made up of structural insulated panels (SIPs). A typical SIP has a core of insulated material such as expandable polystyrene (EPS) with a wall cladding material fixed by adhesion to either side of that EPS core. Panelok manufactures its SIPs in 4 standard widths – 300mm, 600mm, 900mm and 1100mm. Each time a builder sends our factory an order for a building, we produce a 3D model of the building in Sketchup. From this we can export the construction drawings which shows all the internal and external walls of the house broken down into these standard SIP widths; with a few non-standard width SIPs. Typically, over 95% of all the SIP panels in a building will be these standard widths - with only a few &ldquo...

...necessary guide on the system installation,configuration and usage. In summary after the configuration Vtiger users must be able to make outgoing calls, receive calls, and have access to recorded calls. Note: users in crm & freepbx soft phone extensions are already made. Only linking between the 2 servers has to been done in such a way that every crm user's call is routed through the particular softphone extension which is assigned to that user. Also while implementing click-to-call small change needs to be made in the crm which is that the mobile phone field should be masked, last 4 digits of the mobile phone should not be visible to the crm user to increase data security....

I have a working Asterisk 13 phone system, but my Trunk provider (Flowroute) recently retired their POPs and forced everyone to use PJSIP vs SIP protocol. This has literally shut down my phone system. I tried to build a new system from scratch with the latest Asterisk 13 source code, moved over my dialplans to it, etc. but now I can't get any luck with authentication between both my phones and my Asterisk server and the trunk provider. I see traffic coming from there successfully when a call is made, but it is being rejected with a "404 Not Authenticated" response by my Asterisk server. If you are a seasoned Asterisk expert, specifically with experience using Flowroute, this is probably a simple support job. If you are not, please do not bother responding to this...

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 these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

hi my name is kuo ,and my english is pretty basic.. i found this project. https://www.freelancer.com/projects/python/Odoo-Call-Dialler-Click-Call/details use bria5 softphone integrate odoo 13 community. contacts i have this request. I want to use it myself.

I want to develop an android softphone...

Part 1 6 hours The following setups must be implemented within a zoom session, standard working hours 8 am to 5 pm Interventions with down time 6pm to 7am Monday to Friday Installation of a 3CX telephone system appliance 1.0 hours Integrate the SIP trunks for 0.25 hours Connection of SIP phones Yealink T42S Supported 1 hour Fax numbers set up 0.5 hours and Auerswald COMfortel 1200 IP (not directly supported but possible) 3.5 hours Part 2 4 hours Integration of ISDN connections from a Fritzbox - Providing the ISDN connections as a SIP trunk in the Fritzbox 7590 Integration of the Siptrunks in the 3CX telephone system Part 3 10 hours Transfer of all phone numbers and configurations from a Commander 6000 telephone system to the 3CX telephone system - Conversion of al...

This is to call Spain customers to ask for feedback on their recent purchase and offer a discount code on repeat purchase. There will be hundreds or even thousands of customers to call. You need to have good connection to Internet and use softphone APP to make the calls. We will provide the resources. Previous customer service experience is preferred but not a must. We pay by number of calls completed.

Hello Want to develop API fot nat traversal. My client is behind nat firewall. Client (softphone) want to connect to sip server by resolving nat as well as dynamic ip address.

SIP SoftPhone for Salesforce Salesforce integration with the Mizu WebPhone

...while the call between the external number and the ext 100 continues. In some types of failure, a graceful error would be nice. The second internal extension should get the original caller id information passed through as is. I will handle the crm integration. When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address. If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui. Part 2: I need a method to transfer a call to a wav file through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec s...

professional with great experience in Asterisk, PABX, predictive dialer, softphone and wertc

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I am not strong on Python and need help (share screen and give control to VM) Environment: Ubuntu 20 LTS, Python 3.8, Asterisk v16 Note: 1. Asterisk (works OK - Inbound call, plays monkeys via a context with an external softphone) GOAL: 1. A "hello world" python to play the "monkeys" prompt on an Inbound call using Asterisk ARI REST API. 1. Using this Python module ARI 2. See "Hello world example" My basic issue/problem >>> import ari Traceback (most recent call last): File "<stdin>", line 1, in <module> File "<frozen zipimport>", line 259, in load_module File "/usr/local/lib/python3.8/dist-packages/", line 8, in <module> File

Hi there, We need a service of configuring SIP conference bridge for BigBlueButton. Our server is behind NAT, and we can't reach the result - system always says incorrect pin. Would be grateful if you can correct it.