Blackberry sipроботи

...when called. The IVR should facilitate actions like activating a pharmacy pick-up or receiving a response from the software indicating the readiness of the prescription. Clients should also have the option to input multiple prescription numbers successively. Furthermore, they should be able to transfer to a pharmacy technician after entering each prescription. Our company will furnish all necessary SIP information for testing and initializing the IVR. We are looking for a long-term collaboration with the developer, who will not only build these IVRs but also guide our team in deploying such services on a Virtual Machine (VM) in AWS or other recommended deployment methods. The emphasis is on establishing a reliable and redundant service that our organization can depend on consist...

Looking for someone to build a basic flutter Android app for sip client with dart. This will be used for basic call testing with will will build further options with new screens later. For your ref a brief code is attached.

I am looking for a telecommunication engineer who can assist me with VoIP implementation. Specific requirements for the project include: - Expertise in VoIP implementation and SIP - Ability to complete the telecommunication services within 1 week Ideal skills and experience for the job: - Strong knowledge of network design and fiber optic installation - Experience in ongoing maintenance of telecommunication systems If you have the necessary skills and can complete the project within the specified timeframe, please reach out to discuss further is 250$,need to integrate whatsapp with sip, i have written all data i will send.

I am looking for a skilled developer to assist with the development of a Basic VoIP Hosted PBX system to work with SIP protocol with minimal features initially. The project requires the following specific features: -Call forwarding -Call Blocking -IVR -Ring group -Extensions -Call Schedule & Call Queue - Call recording -Greetings - Auto attendant I also need guidance on specific integrations and customizations for the VoIP Hosted PBX. The preferred platform for the development is Asterisk. Ideal skills and experience for this project include: - Proficiency in VoIP technologies and protocols - Experience with Asterisk or similar platforms - Knowledge of call forwarding, call recording, and auto attendant features - Ability to provide guidance and recommendations for integr...

I am seeking a highly skilled canvas artist with excellent social skills to create creative and engaging "sip and paint" ideas. The ideal candidate should be able to produce abstract, landscape, and portrait art pieces. Specific requirements for this project include: - Ability to come up with simple and unique "sip and paint" ideas that will captivate participants. - Strong understanding and experience in working with various painting techniques and styles. - Excellent communication and interpersonal skills to interact with participants during the painting sessions. The client has no specific colors or themes in mind and would like the artist to decide. Therefore, the artist should have a strong sense of creativity and the ability to make informed decis...

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Logo Design for T Party Business example of beginner logo at attached please give us a modern and revamped logo from the example - Name of the T Party business: Sip & Savor - Main color scheme: Pink and gold - Preferred style for the logo: Modern and minimalist We are looking for a talented graphic designer to create a logo for our T Party business, Sip & Savor. The logo should reflect our modern and minimalist style while incorporating the colors pink and gold. Skills and Experience: - Strong graphic design skills - Experience in creating modern and minimalist logos - Proficiency in Adobe Illustrator or similar design software - Ability to effectively use color schemes to create visually appealing designs The ideal candidate will have a portfolio showcasing thei...

We are looking for someone to build a sip client with the functionalities Built with CSIP using Java Configure their SIP account details (username, password, server address, etc.). Register their account with a SIP server. Search for and contact other SIP users. Initiate and receive audio calls. Manage active calls (hold, mute, speakerphone). View call history Use the G.729 codec for all audio calls. This will be a basic demo , to test the features

Hi there i need network administrator for sip to WP integration, i will send details too i have the clear of Asterisk. Let chat to disscus more."Sip provider

I need SIP To WhatsApp Gateway for Converting SIP (session initiation protocol) Voice Protocol RTP Audio to WhatsApp Voice Call Protocol, This also converting WhatsApp Voice Call to SIP Extension / Openpbx (which called "forwarding inbound" external gateway to openpbx extension), for SIP to Whatsapp process we called that "forwarding outbound". Please be noted that this project not allowed to modify the WhatsApp Application Messaging (apk), so we must able to do converting data via Android . Please check the youtube link also

On a really old product server without any OS updates since about 8 years, the A...solutions/scripts and report back debug informations and else. exten => _0049ZX.,1,NoOp(Starting Dialout Procedure app_sound INT ${EXTEN}) same => n,SIPAddHeader(P-Preferred-Identity: <sip:+498936777777@sipxxxcom;user=phone>) same => n,Set(CALLERID(all)=+4917650621344) same => n,Set(__CONF=${EXTEN}) same => n,Set(__INIT_CHANNEL=${CDR(channel)}) ;same => n,AGI(,${INIT_CHANNEL}xxx${CONF}) same => n,Set(FILE(/usr/local/asterisk/var/lib/asterisk/conferences/${SIPCALLID})=${CONF}) same => n,Set(MONITOR_FILENAME=/tmp/wavs/${EXTEN:1}) same => n,Monitor(wav,${MONITOR_FILENAME},i) same => n,Dial(SIP/${EXTEN}@sipout_xxx,30,rG(separator,s,1)) same => n,NoOp("...

We are looking for someone to build a sip client with the functionalities Built with CSIP using Java Configure their SIP account details (username, password, server address, etc.). Register their account with a SIP server. Search for and contact other SIP users. Initiate and receive audio calls. Manage active calls (hold, mute, speakerphone). View call history Use the G.729 codec for all audio calls. This will be a basic demo , to test the features

As per your requirements, I will create an SIP void application in android first and then on ios.

I am looking for a freelancer to create a website for my paint and sip business. The website should have a colourful design incorporating blues and greens. Logo Design: - I do not have a specific logo design in mind and would like the freelancer to create one for me. - The logo should be incorporated into the website. Website Layout: - I prefer a multi-page layout for the website. - The website should have a page for specific events pricing details and FAQ's, a page for policies, and an about us page. Shop Payment Cart and Check-out System: - The website should include a shop payment cart and check-out system for customers to purchase products or book events. Ideal Skills and Experience: - Graphic design skills to create a visually appealing logo and website design. - W...

I am looking for a React-Native (Expo) developer to create a SIP SoftPhone App with advanced calling functionality. The app will NOT need to integrate with any existing systems or APIs. Ideal skills and experience for this job include: - Proficiency in React-Native development - Strong understanding of SIP protocol and VoIP - Experience with building calling applications - Ability to create a user-friendly and visually appealing design If you are confident in your skills and can deliver a high-quality app with advanced calling features, please apply. The app needs to be active in background, just like normal native phone app. And needs to be always avaialble to receive calls. Only thing we will provide are: SIP URI & username/password. Platforms to supp...

Currently I am using VOS3000 Latest Version, I need to be taught how to use it. My client will connect to the VOS3000 using IP and register. and the VOS3000 will set the Route for each client to the Operator's SIP CID. either use one CID or Random CID (Roundrobin, multiple SIP Operator Destination) and set each DID from the SIP Operator to each Client. and how to use a gateway that connects to several SIP providers. and how to use other features on vos3000. a brief overview as in this image.

I am looking for a freelancer who can help me...set up and configure my Twilio account. Here are the details of the project: Setup twilio account for using soft phones Simple IVR menue Sip trunk Preferred Communication Channel: - Both SMS and Voice Call Workflow: - I have a detailed plan that I want to implement with Twilio Technical Level: - I have an intermediate level of understanding with Twilio Skills and Experience Required: - Experience in setting up and configuring Twilio accounts - Knowledge of Twilio's SMS and Voice Call features - Ability to understand and implement a detailed plan - Strong communication skills to collaborate and finalize the workflow - experience setting up softphones/ sip routing If you have the necessary skills and experience, please a...

Hello, I want to work with WhatsApp, the solution should be a web page application based on sip+webrtc.1: User A make a call to whatsapp 2: Wahatsapp receive a text message 3: User A receive ringback 4: Whatsapp click on link and establish the video call are you able to do something like that Regards

I am looking for a freelancer who can help me with the configuration of a Dinstar UC2000-VE device for SIP Trunking TO 3CX. Project Requirements: - Configuration of a Dinstar device for SIP Trunking - Integration with an existing PBX system - Configuration of a single Dinstar device Ideal Skills and Experience: - Experience with Dinstar configuration for SIP Trunking - Knowledge of PBX systems and their integration with Dinstar devices - Proficiency in network configuration and troubleshooting If you have the necessary skills and experience, please provide your proposal for this project.

I am looking for a freelancer who can help me with a project that involves converting SIP voice protocol and RTP audio to WhatsApp voice call. I am open to suggestions and have no specific requirements in mind. I do not have a preference for the programming language, so the freelancer can use either Java or Python, whichever they are more comfortable with. The deadline for this project is flexible, although it would be preferable if it could be completed in a timely manner. Ideal skills and experience for this job include: - Experience with SIP voice protocol and RTP audio - Knowledge of WhatsApp voice call integration - Proficiency in Java or Python programming language

I am looking for a freelancer who can integrate SIP to Whatsapp call functionality for both mobile and desktop platforms. The integration should include basic call functionality without any advanced or customized features. Standard security features, such as encryption, should be implemented for this integration. i need use android as voip gateway. inbound calls via sip protocol into outbound calls via WhatsApp Ideal Skills and Experience: - Experience in integrating SIP to Whatsapp call functionality - Proficiency in mobile and desktop application development - Strong understanding of security protocols and standards

I have a freshly installed "FreeSWITCH", and I want to configure it with the below example: The "Outbound Traffic" - Accept the SIP traffic from the Local "A" () SIP Server. - Redirect the Signaling to the Local "B" () SIP Server. - If the Signaling response is accepted, the traffic (Signaling and Media) will be redirected to the Local "C" () SIP Server. The "Inbound Traffic" - Accept the SIP traffic from the Local "C" () SIP Server. - Redirect the Signaling to the Local "B" () SIP Server. - If the Signaling response is accepted, the traffic (Signaling and Media) will be redirected to the Local "A" () SIP Server. All the traffic mentioned will ...

I am looking for a logo for my photo booth and mobile bar business. My business name is Sip & Snap. The logo should be fun and playful, with no specific colors or themes in mind. I am open to any suggestions. Skills and experience needed: - Graphic design - Logo design - Creativity - Ability to incorporate specific text and symbols

Looking SBC Ribbon Expert Communication between SBC and MS teams - call manager the issue shows codec issue -- i have change it but still same issue.. any one ? I will not pay until the issue resolved ------------ SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/TCP X.X.X.X:5060;branch=z9hG4bK-UX-ac19-5a05-0033 From: "XXXXXXX" <sip:XXXXXX@X.X.X.X:5060;user=phone>;tag=ac195a05-2a;sgid=2 To: <sip:XXXXX@X.X.X.X:5060;user=phone>;tag=44188389 Date: Mon, 27 Nov 2023 13:42:17 GMT Call-ID: call-BD0D1C00-0000-0010-100B-D@ CSeq: 2 INVITE Allow-Events: presence Server: Cisco-CUCM11.0 Session-ID: 00000000000000000000000000000000;remote=00000000000000000000000000000000 Content-Length: 0

I am looking for a VoIP developer who can create a VoIP application for my project. The ideal candidate should have experience in developing VoIP applications with the following functionalities: - Call routing and forwarding - Call recording and monitoring - Voice recognition and transcription I have no specific requirements or preferences for the programming language to be used, so the developer can choose the most suitable language for the project. The VoIP application is expected to be used by: - Less than 100 users - 100-500 users - More than 500 users If you have experience in VoIP development and can fulfill the above requirements, please submit your proposal.

I am looking for a developer who can help me make a connection between my client and a specific SIP server using the PJSIP module and React Native. Preferred Skills and Experience: - Proficiency in working with the PJSIP module and integrating it with React Native - Experience in setting up and configuring SIP servers - Strong understanding of networking protocols and security measures - Familiarity with voice call functionalities and implementing them in mobile applications Project Requirements: - Connect the client's application to a specific SIP server of their choice - Implement basic security measures for the connection - Develop the application to support voice calls functionality If you have the necessary skills and experience, please reach out to dis...

I am looking for a Linphone SIP expert to help me set up new features for social media app. I already have a Linphone SIP account set up and I want to implement additional functionalities like group chat and video conference Ideal skills and experience for this job include: - Strong knowledge of Linphone SIP and its features - Experience in setting up and configuring Linphone SIP accounts - Proficiency in developing and integrating additional functionalities into existing applications Should have hands on experience in setting up linphone video conference..

I'm looking for experienced artist to help my medium-sized team (11-50 people) with a paint & sip activity at our office on December 1, 2023 from 12:30 PM to 3:00 PM to promote holiday fun and the work that we do as advocates for the working class. As it is the holiday season, the activities should be designed to raise the team's spirits and create a festive atmosphere.

Project Description: Sip Trunk user with PBX support I am looking for a skilled professional who can help me with the remote installation of a PBX system. The ideal candidate should have experience working with Asterisk PBX. Requirements: - Familiarity with Asterisk PBX system - Ability to remotely install and configure the PBX system - Proficiency in setting up new features for the PBX system - Experience in configuring voicemail services - Knowledge of call forwarding and interactive voice response (IVR) setup Skills and Experience: - Previous experience in installing and configuring PBX systems - Strong knowledge of Asterisk PBX system - Ability to troubleshoot and resolve any issues that may arise during the installation process - Proficiency in setting up voicemail services...

Hello, I'm in search of help designing a EPS / SIP sandwich panel to be used in the manufacturing of one piece inground swimming pools. The panel would likely use polystyrene foam with steel or aluminum structural members and a metal skin and/or polypropylene skin. I have experience in the pool industry and do not need assistance with any of the hydraulic requirements, structural/mechanical engineering help only.

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I am having a problem with my Blackberry Key 2 phone not being compatible with the RING app. I have not tried any troubleshooting steps yet and the specific issue that I am encountering is that the app is not compatible with the operating system 8.1. I have not reached out to the RING app support team for assistance and so I am in need of someone who can help me figure out what the problem is.

i want to implement a simple architecture. at this moment my asterisk server works fine with endpoints and it can initiate calls in both directions. now i want to use a proxy server in between endpoints and my server . so im using dsiprouter with domain name of and its installed successfully. now there is a problem. i connect my sip users to dsiprouter. and dsiprouter to asterisk as pass thru. when i call any number with sip endpoint there is no problem and the call reaches the dinstar gateway and dinstar call the sim number(ip2tel). but the opposite side of call when sim user call the gateway and gateway route the call through asterisk. when asterisk send request to dsiprouter the request cannot find the endpoint i tried (realm-outbound proxy and etc) but i didnt make it

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I want a small project writen in .net blazor with a SIP Server integration. Server will have sip address username and password. Basic Ui for calling and answering. For demo purpose you can use any SIP server you want. The project will use 3CX SIP server. I suggest to use SIPSocery as library but feel free to use anything you want.

Project Description: I am looking for a skilled developer who can create a WhatsApp to SIP Gateway using Asterisk. The main function of this project is to enable call forwarding between WhatsApp and SIP. I require the gateway to be built on an open-source platform. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. Skills and Experience Needed: - Strong experience with Asterisk and VoIP systems - Proficiency in working with both WhatsApp and SIP protocols - Knowledge of call forwarding and routing techniques - Familiarity with open-source platforms for building gateways - Ability to troubleshoot and debug any issues that may arise during the develop...

I am looking for an expert freelancer to integrate SIP services with mediasoup for my project. The purpose of this integration is to enhance the video calling capabilities on my application. Requirements for the project: - Experience: The freelancer should have expert level experience in SIP video integration with mediasoup This Project may require expertise in VoIP technologies along with WebRTC, Asterisk, kamailio etc. - Tools and Technologies: NodeJS is preferred, however to get the project integration we are open to suggestions and do not have any specific tools or technologies in mind. If you have the required expertise and can provide guidance on the best tools and technologies to use, please bid on this project.

Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated). I already have a small but buggy "solution" that needs to be tweaked. We keep the CID info in freeswitch hash database and collect the data when the parked call is retreived.

I need to install a2billing + freepbx on my server. Create a SIP account and configure a trunk. It must be proven that it works. within 24 hours.

Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated). I already have a small but buggy "solution" that needs to be tweaked. We keep the CID info in freeswitch hash database and collect the data when the parked call is retreived.

Looking for someone experienced with freeswitch/lua and valet call parking. We need to send a sip_ph_P-Asserted-Identity header but it needs to match the caller ID for the call (keep in mind, inbound and outbound calls need to be differentiated). I already have a small but buggy "solution" that needs to be tweaked. We keep the CID info in freeswitch hash database and collect the data when the parked call is retreived.

Good morning everyone, I would need for a customer to massively configure 28 linphone using remote provisoring. Even if only via XML but only configuring the SIP account Can anyone help me?

I am looking for a freelancer who can help me with the installation and setup of a VoIP PABX system for my small scale business operation. Skills and Experience: - Previous experience in setting up VoIP PABX systems - Knowledge of different VoIP protocols and codecs - Familiarity with network infrastructure and SIP trunking - Ability to troubleshoot and resolve any issues that may arise during the installation and setup process

Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between FreePBX, DIDWW, and the extensio...

Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between FreePBX, DIDWW, and the extensio...

...assistance in setting up user accounts, configuring call routing, and implementing security measures. - The configuration should also include integration with other communication platforms, such as SIP trunks and mobile applications. Expected Timeline: - The project needs to be completed within a tight timeframe of 1-2 weeks. - The ideal candidate should be able to work efficiently and effectively to meet this deadline. Skills and Experience: - Experience with configuring Flexisip VoIP servers is a must. - Strong knowledge of VoIP protocols and call routing is required. - Familiarity with integrating SIP trunks and mobile applications is highly desired. - The candidate should have a proven track record of completing similar projects within strict timelines. If you hav...

...an experienced Kamailio & Asterisk VoIP Solutions Architect to design and implement a comprehensive telecommunication solution that integrates WebRTC, SIP users, and robust VoIP services. The ideal candidate will have extensive knowledge in setting up and configuring multi-tenant VoIP systems, ensuring high availability, and developing secure and scalable APIs for system interactions. **Key Responsibilities:** - Design and deploy a multi-tenant VoIP solution using Kamailio for SIP routing and trunk configurations, and Asterisk for media processing. - Implement WebRTC functionality for browser-based communication, alongside traditional SIP user capabilities. - Configure both inbound and outbound call capabilities, including intercom functionalities. - Develop a...

Necesito instalar vicidial + 2 troncales en red local y 2 usuarios para llamar. Deben quedar conectadas las 2 troncales sip y poder generar llamadas las cuentas sip. Plazo de entrega 24 horas I need to install vicidial + 2 trunks in the local network and 2 users to call. The 2 SIP trunks must be connected and the SIP accounts must be able to generate calls. Delivery time 24 hours

This is the work that need to be done: 1. you will setup a Freepbx on a provided with any desk connection. 2. you will setup the SIP Trunk for incoming and out coming calls. (I provide you all the setting data) 3. you will setup the IVR. (simple IVR with different message for opening time and closing time) 4. you will setup 5 extensions 5. you will configure 3 Cisco Phones with the FreePBX, the phone need to be registered and tested for incoming calls, out coming calls, hold, transfer call. 6. you will setup the pfsense with freepbx rules 7. We together do a full test of the system to be sure that is all working.

I need to setup a new freepbx machine, 1 SIP TRUNK, 3 Cisco SIP Phone (You need to be specialist at setting this), 1 IVR.