Sip app blackberrydela

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.

I am looking for a freelancer who can help me with a project that involves using FREEPBX and asterisk ari to place outbound calls out of a sip trunk using pjsip. The purpose of the outbound calls is for customer service. I do not have any existing infrastructure to support this project, so it needs to be built from scratch. For project updates, I prefer communication through email. Skills and Experience Required: - Strong knowledge and experience with FREEPBX, asterisk ari, and pjsip - Previous experience with setting up outbound calls and sip trunks - Excellent problem-solving skills - Ability to work independently and meet project deadlines

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.

Prosimo, vpišite se ali se prijavite za več podrobnosti.

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.

I am looking for a freelancer who can help me set up FreePbx/Asterisk and provision 4 SIP phones for my VoIP phone system. Current Phone System Setup: VoIP phone system Software Installation: I already have the FreePbx and Asterisk software installed. SIP Phones: I am using Cisco SIP phones. 2xCP-6851-3PCC Phones 1xSPA-303 Phone 1xGigaset C530A Skills and Experience: - Experience with setting up FreePbx and Asterisk software - Knowledge of provisioning SIP phones, specifically Cisco phones - Familiarity with VoIP phone systems and configurations I NEED AN EXPERT IN CISCO PHONE.

I am looking for a skilled freelancer to build a SIP Android phone with the following requirements: Features: - Basic calling and messaging features - No need for advanced features like video calling or conference calling - Customizable features based on specific requirements Interface Design: - I prefer a simple and user-friendly design - No need for a modern or sophisticated interface - Custom design is not necessary Timeline: - I need the project to be completed within 1-2 weeks Skills and Experience: - Experience in developing SIP Android phones - Proficiency in Android app development - Understanding of basic calling and messaging functionalities - Ability to customize features based on specific requirements - Strong design skills for creating a simple and use...

I am looking for someone to help plan and facilitate a sip and paint event! The event will feature participants using watercolor paint to create art all while sipping on their favorite beverage. The expected number of participants is 10-20 people and all are beginner to intermediate level. I need the artist to be well-versed and experienced in the watercolor painting technique and be able to guide the participants throughout the event. I'm looking for a professional, energetic artist who is able to create an enjoyable atmosphere for the event and use their creative guidance to help the participants unlock their inner artist.

Hi, I have a SIP Trunk from Saudi Telecom Company (MOBILY) in Saudi Arabia and I want to configure inbound and outbound calls. The SIP trunk comes with static IP address and it's connected directly to the MOBILY company. I need an expert who did this MOBILY SIP trunk connection many times. I need him/her to send me the full configuration steps so I can do it.

I am looking for a freelancer who can assist me with the installation of a SIP license for my OmniPCX Office Alcatel-Lucent system. I need the freelancer to obtain the license as well. Here are the details of the project: Version: We are currently using Version 11.0 of OmniPCX Office. Installation Assistance: I require help with the installation of the SIP license. Number of Devices: The SIP license needs to be installed on 1-5 devices. Skills and Experience: - Experience with OmniPCX Office Alcatel-Lucent system and SIP licenses - Proficient in installing SIP licenses on various devices - Strong troubleshooting skills to ensure successful installation If you have the necessary skills and experience, please reach out to discuss furth...

I am looking for a skilled individual who can modify a Sketchup model of a dome house for panelized construction and engineering purposes. The project involves the following requirements: - Scaling: The main modification needed is to adjust the specific panel sizes in the Sketchup model to reflect a construction using SIP panels with correct measurements and angles. door and window openings are to be accounted for. 2 door openings are to fit a standard 34" door with a rough opening of 36"x83". Ideal Skills and Experience: - Proficiency in Sketchup: The freelancer should have experience with Sketchup and be able to make accurate modifications to the model. - Knowledge of panelized construction: It would be beneficial for the freelancer to have experience or kn...

We are seeking an experienced developer to create a user-friendly VoIP app and an accompanying robust VoIP framework. This solution will allow callers to make internet-based VoIP calls to recipients who lives in an area with out internet, that means the recipient will be using the local network to receive the call. the system has to ensure cost efficiency for the caller. The project requirements include: 1, VoIP Framework Development: Create a scalable and adaptable VoIP framework that facilitates internet-based VoIP calls from the caller to recipients connected via local network infrastructure. This framework should serve as the backend infrastructure for the app. 2, User-Friendly VoIP App: Develop a user-friendly mobile application for Android and iOS platforms tha...

Fusion PBX installation integrated with PSTN SIP trunk, Webphone(SIPML5 or any other for Audio, Video and conference), Webchat and Screensharing configured with Webrtc enabled. on GCP ubuntu22VM Recording Enabled

hi i have run voip example on esp32 lyraT kit and used local sip server(minisip) then it is working fine for call , but i have hosted the asterisk sip server on goolge cloud (the asterisk is working fine as i tested by calling using mobile apps. ) but when esp32 connects with this asterisk server whenever i call from mobile app upon pressing play button it says " no body is available to attend your call" .

Configure flexi sip server for Linphone client. The VOIP call should be encrypted

Please help me connect JIO sip to my freepbx server... Ping ia working perfectly.. Only outgoing is not going to any number thiugh numbers are running and incoming is also not working..

...integrated with above software. 8) Sales funnel with landing page where the data is collected in single place. 9) Add financial calculators with infographics. 10) Watsaap Chatbot for customer care. 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.

I am looking for a developer to build a VoIP web/.net based product for Hotel WakeUp calls, a good example of what we are trying to accomplish is here

Hi I'm looking for an Asterisk AGI written in GO that is probably going to use this library: and which is called f...com/CyCoreSystems/agi and which is called from the dialplan as: exten => 500,1,AGI(gotest,${myVar}) exten => 500,n,HangUp and is able to: * read the 'myVar' variable * read the 'agi_extension' * print to syslog and exit if some variables are missing * execute a saydigit(123) * execute the playback of a wav file * use get_data to get a digit and log it to syslog * set the callerid to 456 * execute a dial(SIP/789) with max ringing 60 seconds and return the ANSWEREDTIME and DIALSTATUS arrays * hangup max bid is 100 euros you must have your own Asterisk setup and GO environment and provide instructions on how to setup and build the ...

Objetivo Geral: Desenvolver um aplicativo móvel em React Native que permita aos usuários fazer chamadas de voz via SIP e chamadas de vídeo usando Jitsi. Todas as APIs necessárias e o servidor SIP já estão desenvolvidos e prontos para integração. Funcionalidades Principais: Dial Pad para Chamadas de Voz: Tela básica com dial pad (teclado numérico) para que os usuários insiram números de telefone. Botão de chamada para iniciar a chamada. Chamadas de Vídeo via Jitsi: Opção para realizar chamadas de vídeo usando Jitsi. Durante uma chamada de vídeo, o áudio da chamada será conduzido via SIP. Controles Básicos Durante a Chamada...

...consumers from the WhatsApp app on their smartphone. We need to develop a SIP to Whatsapp voice gateway to send some of the calls from our PBX. 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. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to th...