Softphones srtptrabajos

We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.

This project is the installation and configuration of fusion PBX on Debian 10. You will be provided access to a VM. What must be delivered Full functional system integrated to 1 sip trunk, VoIP phone and softphones A fully secured system that is only accessible to admin and users registered in PBX Full documentation of how to in Freepbx eg: add new PBX user, configure softphone, setting to be used for softphone, VoIP phone config and settings Auto-attendant with MOH for inbound Leave a VM by pressing a number VM must be sent as an email to admin and PBX registered users. New users can only be added by the admin Outbound calls are restricted to users registered in the PBX. Call recording when a call is answered. You must have experience of doing so as well. The development must ...

...for each call received (see Fig. 1). MapMaker is based on a FileMaker Pro (version 19.2.2) database. A balloon for a marker on the map (Fig. 1) contains a phone number which can be clicked to ring the person or organistion. This is achieved by simply clicking on the phone number. When clicked, a third-party softphone is used to ring the required contact (see Fig. 2). Unfortunately, proprietary softphones contain many more "bells and whistles" than required, evidently cannot provide the desired feedback to MapMaker and are proving to be excessively costly. It is therefore required that MapMaker be modified to avoid the need for a third-party softphone. PROJECT SPECIFICATIONS When a phone number is clicked in MapMaker, it is required that the following parameters be mad...

Configure Twilio For SIP and Configure 2 Softphones and 1 SIP Algo 8201 Device

Dear All, I have a fresh installed FusionPBX on a Cloud [Ubuntu 20.04] and I have configured it to use TLS And the remote registered extensions haven't audio Note: the remote registered extensions are behind a NAT and are located in a country is restricted the SIP traffic so I should use TLS with SRTP to bypass the SIP blockage. Note: Award will be to the one have an experience only to complete this project successful (no wasting time) Thanks to all who will bid on this project!

Hello! I have installed a Cloud FreePBX server [Fully configured] SIP with TLS + SRTP But the user's location is in a country that has SIP Traffic restrictions (Blockage) Everything is working and the calls go through 2 SIP Extensions but there is something wrong with the voice! The voice is like shaking! Note 1: If I use a VPN to avoid the SIP traffic restrictions, the voice is so clear and there is no such problem in the call Note 2: The funny is once I try to dial the other extension of *43 for the Echo test and our voice is still shaking If I pressed any num key twice or three times, our voice becomes so clear! I think may one of the following ideas are resolving the problem: 1- Install & Configure a Cloud SIP Proxy Server like Kamailio, OpenSIPS or any other SIP Pr...

...understanding of network protocols and associated setup required. Skills required: • Hands on experience in JavaScript, Java, C++ etc. • Extensive technical knowledge and working experience in Data transport services, technologies. • Work experience and knowledge in SIP based telephony solution (VoIP). • Working experience on Datacenter, Cloud environment. • Deep knowledge of protocols such as SIP, RTP/SRTP including TLS, webRTC • Very good knowledge of protocol and packet analysis • Experience working with switching and routing protocols • Strong understanding of NAT traversal for VOIP (incl. STUN/TURN) • Experience working with Linux and Windows server operating systems • Knowledge of third party/open source SIP stacks • S...

Hello, I work as a volunteer in an emotional support and suicide prevention NGO. We get about 10,000 calls every day, in which people let off steam for about 30 to 90 minutes. We currently use softphones installed on the computer but I would like to place a SIP proxy and run a webphone in the browser. For that I will need an executable docker-compose file or detailed instructions for: - Frontend - inputs SIP parameters as ip, login and password - button to connect - disconnect button - button to answer the call - button to end the connection - no need to make calls, just receive calls - it doesn't have to look nice, I will adapt the frontend to our internal application. - you can use sipjs, jssip or any other open source SIP library. - backend a backend ...

Hi, I need support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to s...following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to sipgate) - Several Devices with Sip Softphone installed - Home Assistant on several devices (I want to connect via webRTC (Browser) to the Asteriks) So I need: - basic configuration of Asteriks / FreePBX - 2-3 extensions / endpoints for the devices / softphones - 1 extension for the webRTC connection (e.g. ) - 1 configuration for the 2N Doorbell with SIP Support

To set up installed Asterisk PBX stable 16 qnap x86 SSH small set up 4 sip 5 phones 5 softphones GUI - web admin simple voice missed number to email debatable option

I require somebody to take a look at a new VitalPBX install and fix a few small issues I'm experiencing. I've configured trunks, extensions, dial plans etc however I'd like the following things configuring/setting up: - Extensions via TLS - configure device profile for TLS - VitXi - possibility to use push - SRTP working

Hello all, I have Freepbx installed on a VM with a public IP address. I am testing with 2 softphones which can register with PBX and call each other. But there is no audio at all. I would like your assistance to debug and fix the audio issue. I will happy to discuss my setup with shortlisted candidates.

I need a Simple SIP Registration test app for Windows. It will mock an Asterisk server receiving REGISTRATION requests from SIP clients (Softphones or Hard Phones such as Cisco, Polycom, etc.) and it should reply as Asterisk does and it should show the entire dialog and optionally log it to a file. A simple GUI is enough, where I can create an extension (number and password) and then configure a phone to point to it and try to register (the App would get the local IP of the machine it is running on - if more than one is detected, I should be able to chose/select which ones to bind to and listen on). The listening port should also be configurable (default to 5060 but changable). The app should remember the last configuration, screen size and position, etc. I need a standalone / port...

Basic Linphone App with Customization with Own branding App to App calling App to mobile calling Multiple calls management Call transfer, pause and resume Audio conferencing (merge calls into a conference) Instant Messaging Pictures and files sharing Address Book Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications: zRTP, DTLS, SRTP Bluetooth headset support Languages: English, French, Japanese, Arabic Account creation assistant Dedicated tablet user interface Audio  using codecs such as G722,G729,SILK, SPEEX Video calls using codecs such as VP8, H264, MPEG4 Integration with push notification (requires compatible SIP server, sip service has push enabled) Peer to Peer Audio and Video calling using ...

Preciso q o sistema suportar a criptografia dos tráfegos de sinalização e de voz por meio dos protocolos TLS e SRTP para conexão dos ramais aos softphones e ou telefones ips

we would like to implement help desk system / ticketing system Zoho desk/ fresh desk/ bitrix / zen desk or other in our majority of our tickets (about ten tickets daily are coming from phone cal...from phone calls). 6 agents and admin at the beginning. System should have following features: 1. ability to use polish numbers for calls and sms 2. creating tickets from calls, emails, chats 3. recording calls 4. ability to call back direct from open ticket 5. ability to send sms from open ticket 6. ability to send bulk sms to 100 numbers 7. redirecting incoming calls to agents mobile numbers./softphones on mobile devices 8. scalable cloud infrastructure other welcome's features: 9. integration with Microsoft teams 10. crm or integration with crm 11. customer portal 12. time sh...

We are looking for developer to build a softphone Windows, ios, android which needs to be integrate with asterix server. The following mandate features are required 1) Calling inbound and outbound 2 Call Transfer and Blind transfer 3) Call hold 4) 3 way conference 5) SMS Internal & External ( Group and 1 on 1) 6) Video Calling & Conferencing 7) Feasibility to Integrate any othe...mandate features are required 1) Calling inbound and outbound 2 Call Transfer and Blind transfer 3) Call hold 4) 3 way conference 5) SMS Internal & External ( Group and 1 on 1) 6) Video Calling & Conferencing 7) Feasibility to Integrate any other APIS 8) Contact Sync from phone book, email, outlook or 365 9) Users Online/ offline status 10) Geo location 11) Screen Sharing Reference: 3CX, Zoiper ...

I need a softphone (ish) application that can monitor a Sip extension (freeswitch/fusionpbx) display a popup showing incoming call with caller ID number and name. When the call is answered by clicking the answer option on the popup or when answered via the IP desk phone (yealink t46g) I need it to open a url in a default web browser and have the ability to a...the features of their desk phones. They like having a phone, and not having to tie up their computer, they also like having the presence BLF indicators that a softphone app seems to not really provide in a user-friendly way. And they are able to easily and quickly transfer calls, put calls on hold and see who is available or easily send a call directly to a users voicemail. You can do that with softphones but its not very easy...

we are using ubuntu os in the we require the blink software to voice calling . we need support to configuration.

Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.

I need help setting up a FreePBX with softphones using TLS encryption

...developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/programming in Asterisk from Terminal/Linux 5. Resume Looking forwa...

...developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/programming in Asterisk from Terminal/Linux 5. Resume Looking forwa...

...developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/programming in Asterisk from Terminal/Linux 5. Resume Looking forwa...

...developers to join our team. This is for an immediate need and ongoing work. Requirements: - Technicians must handle softphones (configuration, installation, and faults), management of remote help tools, knowledge of redirection and SIP ports, detection of hardware failures. - Developers must have experience on Linux and familiar with coding, VOIP, PBX, Softphone Specialist with Asterisk, FreePBX, Cisco, Elastix, Remote assistance, and Coding Solutions in the cloud. If you think this job would be of your interest and meet most if the requirements, please share with us your: 1. Availability 2. Most competitive hourly rate for an ongoing work 3. Experience with PBX/VOIP and Softphones 4. Experience scripting/programming in Asterisk from Terminal/Linux 5. Resume Looking forwa...

VoIP softphones – mobile and desktop RCS clients RCS Integration

Hi There, This logo is for a Gourment Company, "SRTP Gourment". Company Name is "SRTP GOURMENT" FULL NAME SHOULD BE THERE IN LOGO This company sells spices, Seafood, and many other raw eating materials to be used in the Kitchen. Need a logo to be perfectly seen on a dark and light surfaces.

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Hi Sergey S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. What I want you to do? Create PBX server with SRTP, ZRTP and voice pitcher: I will rent OVH server, so thats not a problem. But I want to have fully encrypted VoIP Calls. Tell me price, and time.

...Javascript, Typescript, Node.js, WebSocket, OpenSSL, Protobuf, pouchdb. 4. DevOps, system integration and support specialist: Total experience 12+ years, experience with o Linux operating system (Redhat, CentOS) o Network architectures and security (Firewalling and IDS) o System architecture for high availability and scalability o Networking and communication protocols (TCP, UDP, TLS, SSH, DTLS, sRTP) o Job automation using Ansible o Shell scripting (perl, python or/and bash) o OpenSSL command line and SSL/TLS certificates management o F5 BIG-IP and load balancing techniques (LTM & GTM) o Deployment and management of systems on cloud platforms like MS Azure, GCS and AWS o IT Service Management Tools (e.g., Jira) o DevOps and in particular experience with Jenkins, GitLab, Git...

About Telp Telp has two businesses one that operates a small call centre with 2 to 6 casuals and another which focuses on being a partner or reseller of Zoho suite of solutions. About the role We require someone who has a demonstrated technical background, and that is proficient in IT systems and solutions, and ideally has experience in setting up VoIP, softphones, telecommunication systems, and most importantly call centre software. We are looking for someone who has expertise to advise in relation especially for set up of software that will most adequately handle remote workers. There are so many call centre software options or softphone options available and we need assistance in evaluating a number of options to consider which would be the best in terms of quality/reliability...

...Javascript, Typescript, Node.js, WebSocket, OpenSSL, Protobuf, pouchdb. 4. DevOps, system integration and support specialist: Total experience 12+ years, experience with o Linux operating system (Redhat, CentOS) o Network architectures and security (Firewalling and IDS) o System architecture for high availability and scalability o Networking and communication protocols (TCP, UDP, TLS, SSH, DTLS, sRTP) o Job automation using Ansible o Shell scripting (perl, python or/and bash) o OpenSSL command line and SSL/TLS certificates management o F5 BIG-IP and load balancing techniques (LTM & GTM) o Deployment and management of systems on cloud platforms like MS Azure, GCS and AWS o IT Service Management Tools (e.g., Jira) o DevOps and in particular experience with Jenkins, GitLab, Git...

Amiba offers leading-edge cloud-based Voice over IP solutions for business. With over half a million MDS Amibavoice communication systems installed throughout Europe, your business will be in good company on one of our next-generation Cloud solutions. Our intuitive unified communications interfaces are available as smartphone apps, PC softphones and desktop keysets. MDS Amiba has been awarded the Editor’s choice for best-hosted VoIP solution, by VoIP Review.

About Telp Telp has two businesses one that operates a small call centre with 2 to 6 casuals and another which focuses on being a partner or reseller of Zoho suite of solutions. About the role We require someone who has a demonstrated technical background, and that is proficient in IT systems and solutions, and ideally has experience in setting up VoIP, softphones, telecommunication systems, and most importantly call centre software. We are looking for someone who has expertise to advise in relation especially for set up of software that will most adequately handle remote workers. There are so many call centre software options or softphone options available and we need assistance in evaluating a number of options to consider which would be the best in terms of quality/reliability...

I want a new asterisk server built to replace our old version. i want someone who will build the server from scratch. Not just copy the sample files and fiddle with them. i require the following functionality. 1. Twenty Extensions. SIP Phone and Softphones etc 2. Conference bridge with IVR to Choose a room. Must support Video and Messages. 3. Short text message between extensions. 4. Video Calling between extensions. 5. IVR with Time of day manu for office and none office. Dial by name eyc 6. voicemail for all users. 7. outbound dial for two factor authentication. needs to call end user. ask question and receive response. (details from database ) 8. 4 Analogue Lines. Digium Card installed 9. 2 SIP trunks. Linked to group of extensions. 10. hunt groups. 11. Connection to Mic...

We are looking to develop a replica of urgently. We dont need all the features from Zoiper. We just need a mobile application (IOS and Android) that can make a phone call directly to a land line or mobile number. Experience in voip applications/ softphones is required.

Hi. I need a push service for asterisk to support Android/iOS softphones.

Hi, We want to add sip phone to our CRM solution. Our CRM solution is web based and developed using by php, node.js, bootstrap. We are still working with some softphones (microsip, eyebam, xlite) by using Asterisk AMI. But we need sip based web phone and it must work with all voip software. We need strong documentation and easy installation. Our OS is centos. Thanks.

...is this is the only point at which the system begins to demonstrate the error (the reason i'm making this post). Our current configuration is the following: 1 - DB instance 1 - Primary calling server A (which is configured to include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) ...

...this is the only point at which the system begins to demonstrate the error (the reason i'm making this post). Our current configuration is the following: 1 - DB instance 1 - Primary calling server A (which is configured to include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server...

We have a PBX running on Asterisk and we have around 60 extensions configured. We use this in order to do remote voice alerts in softphones running on servers and some Axis IP speakers using SIP. We have many Axis speakers working and for this we had forwarded the necessary ports for them to work and we use the public IP address. All of the extensions are outside of the network our server is running. For the softphone we use Jitsi and we have preamplified speakers connected in the speaker plug of our sound card. So when from our office we dial an extension, we have Jitsi auto answer the call and then we speak and call talk and the sound is heard. When we use Jitsi, we use a VPN connexion (OpenVPN) so we use the VPN (local) ip address. But when we work with the Axis IP C3003-E ...

We have Two Avaya SBC Servers with Two SIP lines and 2 IPO Office Ver11 Server edition. We need configure 2 SIP trunks on SBC and Its should work with IPO. Also, We need configure Avaya XI Workspace with TLS and SRTP with Public Access.

We have a problem trying to connect softphones (zapier) to an Issabel PBX. We have the server on our infrastructure and a DID service.

...offer/submission. Schedules for technical staff would vary and would incporate business hours in North America as well as Western Europe. Ideally candidates would have bachelors degree in computer engineering as well as 2-5 years of relevant field experience. Industry accepted certifications would be an asset as well. Candidates chosen would be provided with access to the virtual desktops, softphones and will work closely with the rest of the team on the ground to provide assitance to users and complete required maintenace and project work. Scripting skills are an definite asset VB, Batch, Bash, PowerShell and others. Above skills are the base requirements for any engineers or technicians hired. While we thank all candidates for applying we will be contacting only individ...

...firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself. 2. A publicly available extension, like me@, that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me). 3. Everything secure: No unauthorized access. 4. Be able to switch between RTP and SRTP by changing the configuration. 5. If possible: no asterisk database connection, only flat files. If not possible, explain why. 6. Asterisk comes up and runs as cleanly as possible: no unexplained error/warning messages. 7. Use the defaults that come with asterisk installation where possible. No modifications that will hinder a "pkg upgrade". I cannot grant access to the serve...

Due to hardware problems our very old Asterisk server died on us about 9 months ago, but due circumstances we haven't been able to replac...However we would like to get the situation fixed before the end of the year if possible. Due to the situation the Asterisk will need to be set up on a fairly new server (2x XEON E5-2620v3 6 Core 2.4GHz/32GB(2x16GB)) but with some old(er) peripherals like a DrayTec Vigor 3300V switch which holds the FXO and FXS connections, some grandstream SIP phones and a few regular PSTN phones, as well as Softphones on our laptops and Android phones. Also there is a SIP door intercom we would like to get working again ... We would set up the (Linux) server to specifications we agree upon and the installation of Asterisk and it's configuration would...

... 10. Sms/mms sending receiving 11. Overview of messages by contact / phone number / sip ID 12. Individual message thread 13. Voicemail - listen, record greeting, delete ( downloadable to app when needed from signalwire and deleted in sync) See more : send sms, mms, receive sms,mms midlet, VoIP trial test calling, VoIP softphone skins for both iPhone and android phones, visual voicemail, srtp, use signal wire test account to build a complete solution for one number and ability to add more numbers in a same account or creat more signalwire account seamlessly to work with this app. Open h264 sip video. And all audio codecs for 3G,4g, Wi-Fi data. Call kit integration is a must ( even if a phone is sleeping - app must run in background to revive phone calls and text messages c...

...recent 10. Sms/mms sending receiving 11. Overview of messages by contact / phone number / sip ID 12. Individual message thread 13. Voicemail - listen, record greeting, delete ( downloadable to app when needed from signalwire and deleted in sync) See more : send sms, mms, receive sms,mms midlet, VoIP trial test calling, VoIP softphone skins for both iPhone and android phones, visual voicemail, srtp, use signal wire test account to build a complete solution for one number and ability to add more numbers in a same account or creat more signalwire account seamlessly to work with this app. Open h264 sip video. And all audio codecs for 3G,4g, Wi-Fi data. Call kit integration is a must ( even if a phone is sleeping - app must run in a background to revive phone calls and text mess...

...the Internet Engineering Task Force (IETF). Skype, a popular service, uses proprietary protocols, and Google Talk leverages the Extensible Messaging and Presence Protocol (XMPP). Some softphones also support the Inter-Asterisk eXchange protocol (IAX), a protocol supported by the open-source software application Asterisk." Basic Fetures "A typical softphone has all standard telephony features (DND, Mute, DTMF, Flash, Hold, Transfer, call history, call outcome/disposition etc.) and often additional features typical for online messaging, such as user presence indication, video, wide-band audio. Softphones provide a variety of audio codecs, a typical minimum set is G.711 and G.729." This is a project we would like to kick off right away, there is more f...

User-friendly voice and messaging app with enterprise security. ZRTP vs SRTP No backdoors, no man-in-the-middle attacks Unlimited peer-to-peer calling with out-of-network calling Unlimited peer-to-peer calling Call Mobile or Landlines = VOIP Calling Secure messaging and file transfers Scheduled burn functionality on both ends Available for any Android or iOS device