Blackberry sipKazi

Hi We need someone who has knowledge in SIP , Asterisk to help us set up and outbound system. Please apply ONLY if you have relevant experience. Regards

Looking to configure Audiocodes Mediant 1000 device to forward incoming calls from PRI lines to a SIP trunk outside the premise.

...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...

Full Asterisk IVR Setup training and cdr report with below requirement 1. Creation of IVR flow chart and diagram 2. Communicating with other stakeholders for the Audio files. 3. IVR Configuration on FreePBX 4. Setting up Inbo...stakeholders for the Audio files. 3. IVR Configuration on FreePBX 4. Setting up Inbound routes to point to IVR (via GUI and Command line) 5. Other tweaks on IVR (schedules / time conditions) (via GUI and Command line) 6. Setup of dial plan on GUI and CLI 7. Setup of IPtables and Fail2ban on CLI 8. Install and configure a complete fully functional PBX 9. Troubleshooting 10. CDR Reports formating 11. SIP Configuration 12. Asterisk API 13. Extensions configuration 14. Queue Configurations 15. Webrtc 16..Robo call using CLI 17. Single auto call using CLI 18. F...

I need an iPhone/iPad app. I already have a design for it, I just need it to be built.

If you can create an add-in in outlook and understand SIP/VOIP please reach out.

i need to stop fake ringing on linphone application i used sip account to make a call

Tafadhali Jisajili ama Ingia ili kuona maelezo.

Hello, Looking to get a previously working script modified/updated to work with new underling change. The script was written to work with Asterisk 11 and we are not running asterisk 13 and due a change in code the script no longer works. Phones are SPA5xx The scripts checks for calls that are parted in a specific parking lot and then displays all the related parked c... : 1170 Parking Context : parkinglot_7 Parking Spaces : 1171-1180 Parking Time : 240 sec Comeback to Origin : no Comeback Context : parkedcallstimeout Comeback Dial Time : 30 sec MusicOnHold Class : default Enabled : yes Dynamic : no Parked Calls ------------ Space : 1171 Channel : SIP/RGI-PBX-01-00002791 Parker Dial String :...

Hi, We have a PRI line and connected to SIP server using FreePBX. Currently our employees working from home and using some soft phones but there is no voice clarity. We would like to setup a virtual callcenter which redirect to their mobile numbers. If this is possible please reach me.

Need someone available in KSA Riyadh, We need to configure the Yeaster Part .

I want this, but a dark purple ombre flower in her hair and purple lipstick with a curlier afro.. I want to print it into a clear sticker label.

Hello, I need someone to help us by setting up an inbound call center using Bitrix24 dialer. We have our own sip voip trunk and can also host freepbx vps if needed. We need to achieve the following call flow: Customer calls —> pbx places call on hold while waiting for InGroup member to be available —> once InGroup member is available call is routed to them in the Bitrix24 dialer. Inbound and outbound calls must both work. We need to be able to have multiple agents taking simultaneous calls and multiple customers waiting on hold for InGroup members to take the call. Must be able to play music on hold. We need Bitrix24. I do NOT want anything else such as vicidial or goautodial.

We have a lot of gateways for SMS services. All of them acept AT commands but, only a few have a VOIP access. (SIP protocol). For that, we need an app to convert SIP protocol to AT commands for termination of VOIP calls over SIMs through AT commands.

hello i am looking for someone with freeswitch knowledge person with astpp (billing) will be very good. i need something to do with SIP TLS with letsencrypt and other voip work.. so you need be very good on freeswitch for 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.

I am looking for a fail over to 4/5G when the primary service (Ethernet, XDSL, NBN) fails. This also needs to transfer the SIP calls to the 4/5G backup service. Currently there are a lot of backup services available but i have not come across one where the calls are transferred across. o do this the static IP address will need to be moved to 4/5G service.

Need to install sems-server on Debian server side and Openwrt client side. UAsL <---SIP/RTP---> SEMS-SBC1 <===SIP/MUX RTP===> SEMS-SBC2 <---> UAsR SEMS-SBC2 will be behind NAT.

Need some asstiance getting SIP Trunks setup between Grandstreem UCM6204 and Flowroute SIP Trunk Provider. New UCM and flowroute setup. UCM is functional and able to make calls using FXO. 1. System is registering with Flowroute 2. Assitance with Inbound and Outbound Routes 3. DID routing

Hello, Looking for SIP -> Whatsapp call gateway, it should take the calls from SIP (Freeswitch/Asterisk) and terminate to whatsapp number. Status of number should also monitored on whatsapp that means, if number is active on whatsapp, furthermore if it's getting the rings or not. It should also return the correct call error codes CALL SUCCESS, BUSY, UNAVAILABLE, etc.

Tafadhali Jisajili ama Ingia ili kuona maelezo.

Services: Installation and configuration Install, customize and configure Jitsi meet according to the requirement. Frontend Customization Customize logo, title, font and colors to match our brand Load Balance Architecture jitsi components in multiple servers to meet different load types Video Recording Enable video recording for our conference with jibri installed and configured SIP Integration Integrate our SIP provider to Jitsi with installing and configuring jigasi or let us configure a hosted pbx like Twilio, Voximplant to work with jigasi Autoscale Autoscale video bridges with autoscaling for AWS instances

I need Topex MultiAccess GSM Gateway configuration. Following steps are requied: - Setup SIP Trunk in Topex MultiAccess GSM Gateway and IPX Huawei eSpaceU1960 - Configure outbound and inbound routing in Topex MultiAccess GSM Gateway and IPX Huawei eSpaceU1960 - Testing - Document configuration - Speak Spanish (prefer)

Hi Sir i am doing voip business, i have multiple jobs for you, i read review about you and found you best, 1: Sip dialer that could run in blocked countries 2: - Session Border Controller(SBC) 3: A customized open source voip softswitch with simple GUI

We have deployed 3cx on windows 2019 in our local data center. We need assistance from someone that knows this system. Currently No outbound calls will work, Inbound is only partially working, Phones will not provision (yealink). We are using vitelity as and sip provider. If you do not have extensive experience with 3cx please do not bid.

Hi, I am looking for help with setting up Dail-In and Dail-Out functionality for Jitsi Meet hosted on AWS. Are you comforable and available to provide these services. I plan on using Twilio to get this work done., if you have any other solution please advise on that. Please feel free to contact me if you need any further information [Removed by Freelancer.com Admin for offsiting - please see Section 13 of our Terms and Conditions] or to discuss further.

...chasing a bottle or glass of whiskey. Both should have legs and look like they are running. Must be suitable for screen printing (or similar) on shirts, so heavier line weights are preferred. No cartoon faces, please. The line drawing or illustration should be black and white as it will be used for a single-color screen printing. Meant to represent a "pickleback" where each sip of whiskey or rye is followed (aka "chased") by a sip of pickle juice - sounds gross, but it is surprisingly tasty. Why? Because it is the birthday of a friend. He introduced a group of us to drinking picklebacks AND we all run together - so we want to silkscreen a few "team pickleback" running shirts for the group (he will get his free, of course, as his birthday pre...

1 Kamaillo is already installed, I need Kamaillo configured as a SIP routing server/SBC and data replication both Kamaillo 2. Data replication between both Issabel At the end of the project, I should be able to. , Register sip trunk at the Kamaillo . Register a phone at Kamaillo . Make and receive calls. . Data replication between Issabel PBX . Data replication between Kamaillo SBC

The app is 2 pages (+ login page) Page one: Dial pad showing the credit and the cost per min and remaining minutes + control the call when the user exceed the limit page two: Option for the users to buy credit ( in app purchase ) Spin numbe...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the d...

The app is 2 pages (+ login page) Page one: Dial pad showing the credit and the cost per min and remaining minutes + control the call when the user exceed the limit page two: Option for the users to buy credit ( in app purchase ) Spin number...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the d...

We need a React Native sample app that connects to a sip server and makes audio call. It needs to be like a phone call, using ear speaker proximity sensor etc. Preferably jssip, react-native-webrtc, react-native-callkeep libraries should be used. Interface is not important, we will integrate this sample code to our app.

3CX Sip Integration with CUCM and Alestix

3CX Sip integration with CUCM and alestix PBX

...to develop a SIP to Viber/Bip apps gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/Bip 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 Viber/bip 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 Viber/bip gateway 2) Viber/whatsapp ...

...to develop a SIP to Viber/Bip apps gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/Bip 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 Viber/bip 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 Viber/bip gateway 2) Viber/whatsapp ...

Design, specification and working drawings required for pre-fabricated studio/home office/ school classroom, will also require full specification and working drawings. The structure will be constructed using a timber clad SIP system, i have up loaded similar buildings for guidance, but we would require a unique design.

...servicio. * Desde la pagina de administracion poder pasar con un click al panel del cliente ( sin autenticacion ) * ventana emergente de las llamadas entrantes con datos del contacto, si existe, a la extension. Ya existen soluciones comerciales de este punto, pero necesitamos adaptación a nuestro proyecto. Se trata de integracion con Asterisk, aunque usted no necesita conocer Asterisk o el protocolo SIP, dado que el programa recibirá los eventos de las llamadas entrantes con los parametros necesarios, como tenant y numero llamante ( caller id ). Usted puede elegir el lenguaje orientado a eventos que conozca. Si lo desea, aportamos el servidor para el desarrollo. Sabemos que está desarrollado y se comercializa la solución, pero sin acceso al c&...

Looking for someone with skills in: ReactJS Redux PostGresSQL (Sequelize) Pro in Asterisk and WebRTC Excellent Web responsive developer COMMUNICATION IS IMPORTANT TO ME, GOOD, RELIABLE COMMUNICATION - !IMPORTANT WEBRTC WITH SIP via SIPJS, JANUS, or SIPML5 - !IMPORTANT SKILLED AT MAKING DIALPLANS IN ASTERISK AMI AND ASTERISK AGI - !IMPORTANT NEED SOMEONE WITH VERY GOOD ENGLISH - !IMPORTANT The application is in the process of being built. I need someone to take care of the additional tasks. I need someone who can dedicate their time and provide clean and organized code. 1. "Build in call routing into the groups. Inside the SMS Group settings, when a call comes in we need to be able to decide if the call will: - Be forwarded to ring the last agent that spoke to the customer w...

CRM company looking to add a multi-tenant PBX solution. Currently using Asterisk but looking for something better to handle many simultaneous customers and to API with current software (php, mysql, ajax). Require the ability to add, edit and remove SIP users along with codecs, voicemail and call history. Will use Twilio for origination and multiple wholesale providers for outbound. The engineer awarded must create the AWS linux instance, configure PBX and assist in providing API information. Additional maintenance work will be needed.

calculate load from one SIP wall with external finish. full structure design including beam has already been done. Just one sip wall load needed.

SIp client for presence, registration, chat, audio call, video call, and group audio/video call.

We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server

Kiosk mode, SIP softphone in C or Java for linux as per the pdf attached Note: any paid/third party component would not be acceptable.

I want to create a sofphone for Windows using Linphone SDK Softphone should be able to dial direct from GUI and I should also be able to send command from other application to dial X phone number Program must be nice GUI and I will need source core with all libraries once paid in full.

Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 7. Server A to Server B voice traffic will be e...

Asterisk 17.4 is downloaded and extracted, but needs to be properly installed onto a server for us. PJSip config to a SIP provider, the ability to provider music/advertising on-hold, a couple simple queues. We do not want freepbx, please do not suggest it. Just install asterisk and, as needed, help get config files setup.

We are a telecom startup (with existing customers in beta and ready to launch soon) and we’re looking for a PHP full stack web developers with experience in twilio and having some knowledge of telecom, not 100% required but would also be great if you have knowledge with FreeSWITCH & OpenSIPs (both open-source telecom software & PBX / SIP Server). You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line Ubuntu/Linux/SSH MySQL GIT Twilio These skills are optional: AWS Facebook APIs React Telecom knowledge FreeSWITCH OpenSIPs We will offer long term work every week but only if you prove your skills in the first week. We are only looking for serious developers and to prove that please fill the attached docum...

We’re developing a business to build and sell back yard offices in the US. The idea is to build three variants: 64, 96 and 112 sq ft. The materials specifications we’re looking for: - Walls, floor and roof structure: 4” PU SIP panels. - Double glazed timber door system, fixed side lights and top swing window. - Roof: single ply flat roofing membrane. - Cladding: vertical red cedar cladding left to weather naturally. - Vinyl flooring. - MDF painted skirting. - Walls fully plaster boarded. - Lights: 4 chrome LED lights inside. 3 outside. - RCD breaker. - PVC gutters and downpipes to rear of pod. - Some illustrative pictures are attached. What we need is a professional with experience in the construction business to help us with: (1) Validate specifications with ou...

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.

Install Asterisk in the same 3cx Server make a General SIP trunk with DID number from one of your internal extensions, then add to this trunk DIDs for all of your internal extensions; You need to add inbound rule for each of the trunk DID and configure it to route calls to corresponding extension number. Integrate on Vtiger and zoho CRM , noting that Vtiger is on cloud server and thus the Vtiger and Asterisk Server are different location, then MySql need to be open to connect Vtiger and Asterisk.