Blackberry sip softphonepekerjaan

I have a voip project need use the LinphoneSDK + Flexisip server. I have setup the Flexisip server but I found if we use 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 miss. also there one more thing about the mysql database authentication.I try to enable the mysql database authentication. But it not work. The file storeage authentication works fine. I need some developer with the experience about the LinphoneSDK + Flexisip server to help me. Thanks Frank

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.

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 VP...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 Modem. 6) Prepare complete Document and COngiration steps and document for our record. 7) reset and factory default setup to ensure configuration remain there. 8) Configure the GOIP Device as well. 9) Connect the GOIP device with SIP Server to make calls. we have lcoal and Cloud both Server to make sure GOIP Is Connected to CLoud is impo...

Hi 1- Fix odoo softphone and add udp connection to it if possible . 2- Create calls manager module to connect to Issabel pbx server through AMI ( Asterisk Manager Interface ) .Or edit existing module but with direct connection without using any middleware software like odoopbx for example .I have pdf file also for requirements .

Hi 1- Fix odoo softphone and add udp connection to it if possible . 2- Create calls manager module to connect to Issabel pbx server through AMI ( Asterisk Manager Interface ) .Or edit existing module but with direct connection without using any middleware software like odoopbx for example .I have pdf file also for requirements .

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 with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

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 with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/pr...

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 with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

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 with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/p...

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 to be able to automatically switch to the backup in case of failure. We need the freelancer to suggest a PBX (open source if possible), install configure and document the read the description carefully before bidding.

I need you to recommend me a call center platform. I need a script upload that is easy to use along with a softphone hooked to the account

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

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) No Time waster please. Bid only you have knowledge on JITSI Meet Configuration with those. Thanks, Ranjit

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 for learning. I can provide the credentials for testing if it connects to my asterisk server. App should be able to register, make call, and hangup. This will require WebRTC. No need to worry about trunking as my asterisk server will take care of that. See screenshots for designs.

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]: member.c:389 member_exec: unable to answer call [Jul 16 20:35:28] WARNING2774[C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel if someone help me to make a call for french phone. Thank you.

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 pstn can connect to the jitsi instance 4) Connect our jitsi instance to google transcription. I will provide the API key

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

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 avatar character in as a video participant, allowing us to control the lip movement we do via api. So we thought maybe a screen share, where it is a website which auto plays the avatar or if not, we would have to send over mp4 video files to the solution for each prompt the avatar says. see this type in a question and see how it responds, we want this same thing, but not have

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 avatar character in as a video participant, allowing us to control the lip movement we do via api. So we thought maybe a screen share, where it is a website which auto plays the avatar or if not, we would have to send over mp4 video files to the solution for each prompt the avatar says. see this type in a question and see how it responds, we want this same thing, but not have

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 and blogtalkradio.com. and and with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. THESE ARE ALL THESE SAME THING WITH JUST A DIFFERENT LOOK so if you think they are different then you dont know what you are doing so keep moving along. NO PREPAY . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more. DO NOT BID IF YOU WILL TELL ME YOU HAVE VIRUS, YOUR relative is sick or

Hi I would like a module that...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: The Module need to be able to: == Front end == Self Service Interface Sip Trunk Dynamic Dashboard End User Portal Call History log Coverage More info == Back end == Modular Widgets Statistic Widgets Kunden Invoice für DID Reports and statistics Coverage Management Custom Call Forwarding Options Rates and Pricing Setup Payment modules Anti-Fraud Currency configuration Database Wipe/Export/Import Please feel free to ask for more info.

...for special occasions and grandma’s go to for common colds to stomach issues. And today, after countless disagreements with my two children on making better beverage choices, they marvel at the thought that a simple flower, in our backyard, could be transformed into Meila’s Hibiscus Tonic. Named after my daughter Ameila a.k.a Meila, this tonic, is as bold in taste and flavorful as she is. Sip it hot or cold and it may just improve your well-being. This caffeine free beverage is an immune system booster, a great source of antioxidants, may lower blood pressure, aid in weight management, reduce blood sugar levels and treat anxiety. Cheers to a reinvigorating well-being! At Meila’s • We strive to seek the best of the best from all over the globe. T...

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 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.

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.

2 families just bought a farm for weekend recreation and we are naming it the Blackberry Acres Ranch or BAR (what are you doing this weekend - spending it at the Bar) and are looking for a logo we can put on t-shirts or can cozies or business cards. We have a ton of Blackberries on 66 acres and are Planning on having some bee hives out there. I’m attaching the layout of the land and some pictures in case that helps. And it is located in the town of Cuba right next to the town of Bourbon if that is meaningful. Our last names are Oswald and Etchason so throwing an O and E somewhere are totally inbounds. Or Oswetchason which is our ship name. As for theme, something country/rural would be good but don’t really want it to be hokey.

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.

...(both Mac and PC), phones and identifying random cables and power plugs to help me sort through my 10 years of "tech" stuff in my house - and basically transfer the data if any, somewhere else online e.g. iCloud or Dropbox, and then wipe and destroy all my old devices. This is not an easy task as there are tonnes of cables and I don't know what goes with what and some of my old phones (e.g. a blackberry) - I have no idea where the cables are. It will be helpful if you have some personal "master" cable. The devices I have are: - a very old Mac - a very old MacBook - 3 old windows laptops (I think Toshiba, Asus and one more can't remember brand) - about 5-6 old handphones And all of the above - needs to be data transferred, data wiped and then ...

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.