Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.
...developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development u...
...developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development u...
Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.
I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50
We are seeking a skilled freelancer with expertise in Asterisk to assist us in implementing outbound call functionality to a mobile number using PJSIP. The current channel originate command is not working. and we are open to alternative methods that successfully initiate calls. Earlier, when we used older version of asterisk 18.1, we can get calls but newer version doesnt have that command and it gives below error pi*CLI> originate sip/ application Playback hello-world No such command 'originate sip/ application Playback hello-world' (type 'core show help originate sip/' for other possible commands) We are using asterisk 20.5 and system is rpi with ubuntu 23 installed. Thank you!
...We prefer Twilio. But are not sure if Twilio is offering a Sip to Whatsapp Gateway service - You will build the script / library that enables asterisk to terminate calls to a whatsapp endpoint. - Asterisk is build from source. - A AWS Vps to work on can be provided Here is what the dial plan code could look like that needs the extra option for whatsapp calls same => n,Dial(PJSIP/+${outboundNumber}@${trunkprovider}&PJSIP/${whatsappNumber}@${whatsappTrunkPriver},30) Deliverables - Documentation on how to build asterisk from source with extra libraries etc - A proof of concept that it works. Ideal skills and experience for this job include: - Strong knowledge and experience with Asterisk call termination - Experience with integrating Asterisk with WhatsApp - Familia...
Complete Solution for GoIP Hardware (GSM Gateway + SIM Bank) Budget: More than $1000 Primary Purpose: Automation The main goal of a project is a creation of software platform to manage a number of distributed network of GoIP GSM gateways and SIM Banks (SMB). We should be able to make mass calls, SMS/USSD bulk sending & receiving. Asterisk PBX must be used as...SIM/channel enable/disable/restart - IMEI display/modify - signal level display - human behaviour simulation (to increase SIM cards lifespan) - bulk SMS/USSD sending & receiving - automated SIM cards check, kick dead (blocked) SIM cards if detected - live tracking of ASR and ACD for each SIM card and channel - live total / answered / failed calls per SIM/channel Asterisk part must be built on top of PJSIP and ARI (...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHN...connection is made, then send messages between some clients and then terminate the app. This is just the basic summary. Please refer to the PDF attached for the full specification. I don't want to use WebRTC due to its high latency, but instead need to use raw UDP packets (SOCK_DGRAM). This is both portable and performant. There are various NAT traversal libraries out there, but I prefer using PJSIP because Android and iOS devices are supported. If you prefer to use another library, please consult me first. Thank you CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY ...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHN...connection is made, then send messages between some clients and then terminate the app. This is just the basic summary. Please refer to the PDF attached for the full specification. I don't want to use WebRTC due to its high latency, but instead need to use raw UDP packets (SOCK_DGRAM). This is both portable and performant. There are various NAT traversal libraries out there, but I prefer using PJSIP because Android and iOS devices are supported. If you prefer to use another library, please consult me first. Thank you CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY ...
CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY BACKEND WORK! DO NOT USE CHATGPT IN YOUR PROPOSALS. YOU WILL BE IGNORED. Hi I ...to a list of backup TURN servers. Once the connection is made, then send messages between some clients and then terminate the app. I'd rather not use WebRTC due to its high latency, but instead prefer using raw UDP packets if possible (such as sendto, recvfrom etc). Is this something that can be done in C++? I notice that CoTurn has a library you can use, but there are also other ones such as Pjsip. CANNED RESPONSES WILL BE IGNORED. DO NOT MENTION IN YOUR PROPOSAL THAT YOU HAVE SKILLS IN IRRELEVANT TECHNOLOGIES SUCH AS PHP OR CSS. THIS IS PURELY...
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 discuss fu...
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
We are seeking an experienced freelancer to facilitate the seamless integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the u...capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the utmost security for our call traffic through Encrypted Media. Project Requirements: -Integration of WebRTC functionality into the Isaaabel platform. -Configuration of JsSIP npm package to establish connections with our Asterisk calls to SIP and PJSIP extensions through the WebRTC interface. -Implementation of robust security measures, including encrypted media, to safeguard call communications.
I am looking for a freelancer to help me integrate a WebRTC based calling experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement s...Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement so it would be great if the freelancer can deliver the project quickly. Resume, we need to connect WebRTC extension with Issabel. I need that JsSIP npm package can connect with my asterisk server and make calls to SIP and PJSIP extensions. I need the documentation of the implementation for future re-install ...
I am looking for a freelancer who can help me deploy Asterisk 16 with a specific PJSIP configuration. The ideal candidate should have experience with Asterisk and PJSIP. Requirements: - Familiarity with Asterisk 16 - Ability to configure PJSIP according to specific requirements - Experience in handling concurrent calls, with a focus on optimizing for a single call The requirements are very simple. I have configured it myself before and achieved single-pass. If the extension calls the mobile phone, the sound of the mobile phone can be heard, but the sound of the extension cannot be heard by the mobile phone. You only need to configure the phone to be able to achieve dual communication! If the price is not suitable, the price can be negotiated as long as you can solve...
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
Hi I will share the details with the shortlisted candidates. Thanks
I want to add a VoIP client into our app to make and receive phone calls. Looking for iOS and Android developer To be able to send and receive SIP calls to our Freeswitch server I want to use something that doesn’t tie us to a CPAAS platform that requires us to buy phone numbers from them. Since we have our own VoIP sys...VoIP client into our app to make and receive phone calls. Looking for iOS and Android developer To be able to send and receive SIP calls to our Freeswitch server I want to use something that doesn’t tie us to a CPAAS platform that requires us to buy phone numbers from them. Since we have our own VoIP system .So something like pjsip -No need for registration- -no need for incoming calls- i just care about outbound calls for now Please note about your exp...
OpenStage 20E SIP Firmware I am looking for a skilled professional who can assist me with installing the latest version of OpenStage 20E SIP Firmware. Requirements: - Experience with OpenStage 20E SIP Firmware installation - Familiarity with Asterisk PJSIP Specifics: - The current version of the firmware is not specified, so the freelancer should be able to handle any version (1.0, 2.0, or 3.0) - The main goal is to ensure compatibility with Asterisk PJSIP - The project requires immediate attention, so the freelancer should be available to start working on it right away Deliverables: - Successful installation of the new firmware - Verification of improved audio quality, enhanced security features, and bug fixes/stability improvements If you have the necessary skills and...
...Additional requirements: - Integration of Fail2Ban to enhance security and protect against unauthorized access - Implementation of PJSIP for improved audio quality and performance - Do not use FreePBX or any other Bloatware. Your solution will be rejected and you will not be paid if you do that. Has to be done only via asterisk. - Demonstrate working inbound voicemail service that plays a greeting and accepts voicemail on Telyx SIP lines. Ideal skills and experience: - Strong knowledge and experience with Asterisk and Ubuntu Docker - Familiarity with voicemail systems and call routing - Expertise in integrating Fail2Ban for security purposes - Proficiency in implementing PJSIP for enhanced audio quality - Asterisk has to run in Docker with a specific IP address in the s...
...Additional requirements: - Integration of Fail2Ban to enhance security and protect against unauthorized access - Implementation of PJSIP for improved audio quality and performance - Do not use FreePBX or any other Bloatware. Your solution will be rejected and you will not be paid if you do that. Has to be done only via asterisk. - Demonstrate working inbound voicemail service that plays a greeting and accepts voicemail on Telyx SIP lines. Ideal skills and experience: - Strong knowledge and experience with Asterisk and Ubuntu Docker - Familiarity with voicemail systems and call routing - Expertise in integrating Fail2Ban for security purposes - Proficiency in implementing PJSIP for enhanced audio quality - Asterisk has to run in Docker with a specific IP address in the s...
I am looking for a skilled developer to create a custom VoIP Autodialer software with the following features: - Automatic call scheduling: The software should have the capability to automatically schedule calls based on pre-set criteria. - API to let any 3rd CRM integrate with us (e.g. push leads to Autodialer) - Asterisk-based system built on top of modern PJSIP-stack w/ real-time DB - Real-Time Web UI utilizing websocket technology I have a specific preference for the programming language, which is Python. Can be Node.js also. The ideal candidate for this project should have: - Proficiency in Python or Javascript (Node.js) programming languages - Experience in developing VoIP applications - Knowledge of CRM integration - Familiarity with automatic call scheduling - Experience ...
We are working on a video project through PJSIP for IOS & Android, and we need a senior expert who has worked with those kinds of projects/tasks so he/she can help us how to show my own cam video on the video call
I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode
I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode
I want a video tutorial on how to install PJSUA2 on Ubuntu and/or Windows for Python. The video tutorial has to work. I keep getting errors initializing the script.
Hello! I'm looking for an experienced developer to configure lib open H.264 for a quality video stream video call on a Mobile platform. The programming language I am looking for is C++, The perfect candidate will have extensive experience in this particular field and the ability to create a stable and high-quality video stream. - Im using PJSIP lib to make VIDEO CALL - Lib Open H264 config here: - I need modify function oh264_codec_open with best param for mobile device to show HD VIDEO Currently i can able show HD Video but not optimize about CPU & Quality , VIDEO show lag and like image bellow when network slow or network fast will show sometime. I need guys who have
...this topic. Expected solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to help us to build the evil side ;-) MAC (we provide linux+windows) In the case you have to build a dll of pjsip for macOS x64 and ARM, along with the dependent libraries. This libraries have to be loaded via java 17+ Examples of required work: ) So the app has to work on a desktop (macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working sip account after
I need help building newest PJSIP library 2.13 with support with TLS. - Support full codec -- sample IOS project Swift code - Need someone with prior experience building open source projects and with PJSIP.
...solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to help us to build the evil side ;-) mac (we provide linux+windows) In the case you have to build a dll of pjsip for macOS x64 and ARM, which detects incoming SIP/VoIP calls and take over tasks managed in a issue tracker for the java, c++ side and for linux, windows and mac Examples: ) So the app has to work on a desktop (macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working
...focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to help us to build the evil side ;-) windows (we provide linux) Ideally you have also mac experiences too In the case you have to build a dll of pjsip for windows, which detects incoming SIP/VoIP calls and take over tasks managed in a issue tracker for the java, c++ side and for linux, windows and mac Examples: ) So the app has to work on a desktop (windows, linux, macOS) and have to communicate with a SIP provider only. e.g (we will share you
...right partner for onging work on this topic. Expected solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to make a self executable app (in Java), which detects incoming SIP/VoIP calls. On incoming calls, pjsip (c++ lib) app opens a browser with a (caller-)URL (details see below) Examples: ) So the app has to work on a desktop (windows, linux, macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working sip account after award)
...right partner for onging work on this topic. Expected solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to make a self executable app (in Java), which detects incoming SIP/VoIP calls. On incoming calls, pjsip (c++ lib) app opens a browser with a (caller-)URL (details see below) Examples: ) So the app has to work on a desktop (windows, linux, macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working sip account after award)
...access web interface - Billing System for subscriptions that integrates with our credit card processor, Stripe - Reseller Support - Auto Dialer for Client Call Centers, with admin interface and exportable logs, easy import for dial lists. - Automatic Failover and Load Balancing for Redundancy across multiple physical servers / datacenters Current Features we do have: - SIP/PJSIP - Find-Me/Follow-Me - Voicemail to Email - Call Waiting - Call Whisper - Do not Distrub - Extension to Extension Dialing - Call Forward - Call Parking - Call Transfer - Caller ID Screening - Successive or Simultaneous Ringing - Three-Way Calling - Voicemail - Custom Hold Music - Dial by Name Directories - Multi-Company Profiles for Cli...
Hi , i want to add video feature call into this lib wrapper pjsip My requirement: - Create project sample - Import and use above opensource - Add video features follow document of PJSIP - Document result & comment code when finish project
im using Pjsip to make call i want to add features to it support: - Video call ( i have add some class but it not enough, not try, you can see in Omikit Class on repository of you ) - support: call Video fullscreen like sample video - toggle camera on/off - Mute mic - can show dialog choose sound device - can switch camera to front/back The project sample with UI like this
Hi friend, I have source code Look in Pod dependencies this source using Opensource library I want rebuild that library with - newest PJSIP library 2.12.1 - Support codec Opus Run build by goto source Vialer-pjsip-iOS and run command: ./vialerbuild --pjsip-version 2.12.1 --enable-ipv6 --opus -h264
Old Config ; -COPYLINK exten => 441303850708,1,Dial(PJSIP/441303720086@sip-provider) exten => 441303850708,2,Hangup() ; -KENT SCHOOL UNIFORM exten => 441233813666,1,Dial(PJSIP/441303720086@sip-provider) exten => 441233813666,1,Hangup() ; -Vonage exten => 441303766206,1,Dial(PJSIP/441303720086@sip-provider) exten => 441303766206,2,Hangup() ;HOMESTART exten => 441233646709,1,Dial(PJSIP/447593922095@sip-provider) exten => 441233646709,2,Hangup( Config on the new PBX should be the same.
Hi friend, I need build Opensource library with - newest PJSIP library 2.12.1 - Support full codec - Support ipv6 - sample IOS project Swift code use this lib to connect our sip platform ( will give account demo ) Support both ipv4 & ipv6
I need live support for 2 hours to show and fix in my pjsip android project
I would need to Integrate Vtiger CRM 7.X and Issabel PBX. functions NEEDED on the Vtiger int...CRM 7.X and Issabel PBX. functions NEEDED on the Vtiger interface: - popup with caller information (only for the agent who answers) - click2call - History (including recording) What will I provide: - a fully working VPS - vtiger and Issabel (Asterisk) already installed on the VPS Important additiona informatins: - All the extensions MUST be PJSIP - I have a working version of the integration between vtiger and Issabel, but it doesn't work with PJSIP extensions. In case you are interested into working on this version, just let me know! - If you plan to customize the Asterisk callflow, you have to warn me PS: For those who don't know issabel, it is a Freepbx fork. ...
i want a python expert who is familiar with Pjsip . ASAP. further detail will be shared in chat box
i want a python expert who is familiar with Pjsip ASAP. further detail will be shared in chat box
