Trunk sip asterisk quintum tenorproiecte
Hello. I have one physical device and 3 soft phones that I need configured with twilio sip. Would you be interested in this project?
Edit: Please use Times New Roman as the logo font. --- Please make a logo similar to the images supplied. I reckon using the outback photo as backgrounding over the whole map might be a step in right direction; and putting the business name towards lower right-hand corner without covering the trunk of the small tree. Final file required is a vector file, black and white, colour, And a transparent background.
Overview: The Voice changer for Asterisk represents DSP software solution that is able to change voice of a participants in two way voice communication. This is an Asterisk module that implements an asterisk application that can be used from dialplan. It should be able to change human voice of a speakers in a quality way. It should be able to manipulate voice attributes like: chiefly frequency, harmonic structure, intensity... It also must be capable to identify and change formant of human voice. Resultant real-time voice must have the following characteristics: it is clear, it keeps intimacy of a speaker and it is natural. DSP should be able to produce at least 10 different resultant voices from a single speaker. It should be possible to set all parameters that define...
Hello, We are developping a softphone/webphone like zoiper. We developed user interfaces and now we are looking for someone for developping functionnalities like : SIP account registration Receiving call Calling out Transferring call On hold call Recording call ... Best regards,
Mobile application for Android, iOS and web (preferably react-native or flutter), which can authenticate against an internal IDP (identity provider), then based on the SIP credentials (SIP URI, username and password in the auth payload), the application should be able to join the SIP call by sending an invite. This is a very high level overview, the application should allow following features - login (using an IDPm such as pingidentity or auth0) - negotiate a token for validating the SIP server - search of existing users and conferences. - join using SIP client built inside the platform Our infrastructure is in GCP and we would need you to work closely with our GCP infra as well, so knowledge of GCP will help.
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
I would like a logo design for my company quantum projects. I have the basic idea of what I want but would like someone to refine the idea. The logo consists of the name "quantum projects" in lower case. the q and p form the ears of an elephant. The letters q and p share a vertical line that is effectively the 'trunk' of the elephant. The trunk of the elephant tapers to a sharp point to symbolize progress towards and objective. The tusks of the elephant are half-circular arrows, almost like the 'spin' of entangled electrons in quantum computers. Please refer to my rough draft of my initial idea. I will select the entry that makes the most of this idea and is most refined and makes the best choices in terms of colours, fonts and lay...
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...interfaz WEB: Se requiere el desarrollo de una interfaz WEB que se conecte mediante la API a Asterisk para la obtención en tiempo real del estado de las extensiones y comunicaciones. • Información Extensiones y su estado en tiempo real. La interfaz web también dispondrá de la posibilidad de conectar a los ficheros XML de configuración y permite su edición de forma gráfica.. • Información de llamadas. • Mostrar LOGS • Alta de EXTENSIONES • Alta y gestión de CONTACTOS NOTA: La plataforma se encuentra completamente operativa y funcionando. Todas las funcionalidades a nivel de voz y servidor ya se encuentran desarrolladas. Tecnologías deseadas: • JAVA • Fr...
I need an voip sip Android app. I already have a design for it, I just need it to be built.
Set up cloud based Install of elastix v4 in AWS instance Setup Elastix 4 firewall, iptables, fail2ban, etc. Setup trunk and 5 extensions.
We require someone who has experience using / programming in Asterisk to provide a white labeled app for use on Iphone & Android. We require all features as noted here: We also require call recording (to be stored on server or customers server) as well as Caller ID forwarding.
Need a simple Zello to SIP interface that will allow a phone on an Asterisk PBX to call into a Zello channel. When talking on the phone it will transmit/receive over the Zello channel.
We have a freepbx server, and we need to make calls based on csv files. We may can use our server, or maybe another third part service, if suggested/approved.
Hello I need to setup the Bitrix24 telephony system. Attached to 3 different SIP servers. Thank you
60 to 70 percent work is already done on freepbx but something's still incomplete in my project. First i want to explain what is my requirements. I want a asterisk base caller/callee filter solution on PHP which can filter incoming traffic and create whitelist and black list base on condition of repeated callee/caller and non repeated callee/caller with VoIP call bandwidth optimizer. It will also work from 1ip to another ip.. Blacklists and whitelists of phone numbers/ Caller IDs on the basis of repeated numbers/caller IDs. Maintaining Call Gaps between calls to simulate HB Monitoring number/Caller ID length pattern while receiving calls Optimizing Bandwidth Efficiently
...the number and duration of calls. -Counters to reset at the specified time. -Live statistical data on the performance of your SIM-cards. -ASR and ACD for each SIM-card and channel -Channel performance status. -RSSI (signal strength) of each channel. -The duration of the current call. -Activity and statistics for each SIM-card. -Rapid notification of problems via Skype, SMS or email. 5: Built-in SIP server to work with SIM card bank. We can show you the cases of the similar services which work with GoIP equipment so you can get better idea....
...good tree shape but has too many sharp angles. There is also an issue with the branches looking like a tree monster with eyes. Example C - This design is too complex and the points on the branches draw too much attention. We want the complexity to be somewhere in between Example C and Example A. Example D - This design has the Australian tree shape that we like. We do not like the way the tree trunk is bending and feel that the detail of the leaves are too fine. Example E - An example of Australian looking leaves. Notice how the leaves are longer and have a slight curve compared to Example B. This is optional as the best design might have the leaves of the tree represented as more of a block or shape of colour....
Looking for someone that knows asterisk and knows how to code in C. Looking to fix chan_alsa to support sending calls to more then a single alsa device including virtual devices from within the dialplan. The idea is the virtual devices will be setup to stream to AES67 receivers and allow overhead paging. Anyone that also knows AES67 is a major plus
I need to connect an 8 channel DINSTAR gateway to my asterisk server and create extensions to configured and ip iphone.
...the following project. Support for EVS audio codec 1 Introduction A .net DLL and/or Com DLL to support decoding of recorded audio call with EVS codec. The idea is to build a .net DLL and/or Com DLL which can run on Windows (x64) platform and capable of decoding recorded media using the EVS codec in which it was recorded. 2. Key information PCAP file has all RTP/SIP for a call and will not contain any other packets. Only SIP and RTP are related to one call. Sampling rates- 8, 16, 32, and 48 kHz Modes & Encoded Bandwidth- NB (20-4000 Hz) WB (20-8000 Hz) SWB (20-16000 Hz) FB (20-20000 Hz) 3. Requirements o The end software should be a .net DLL and/or Com DLL. o C++/C# Source code of the dll created. o It should Expose a few methods/functions to use. o The main decr...
I need a little customization on Issabel Call Center Community Edition. I just need to send the calls with some delay to the SIP Trunk, lets say, half second of delay on every call originated by the dialer. The goal its to reduce the CPS load on the trunk. I need someone with skill to get the job done! Its urgent!
Hi, I have a phone API to built. You must gather all the information that the API could offer, such has retreive client phone call. We need an angular pho...check our DB is the client exist, so we need a small square that will show us the client information on the call we receive and once we press the button we will be redirected to the client file. We have a CRM that we use, so you will have to create a module for that. So you need to have experience, your code will be double check for the quality of it. You need experience with CI, PHP, Angular, SIP protocol and you need to be able to understand the API has the company does not offer any supprt. I will provide informations, picture and link for you to complete the work. Come to chat I will anwser all your questions. Good l...
delay the buy signal on asterisk or opensips, when the call is hangup delay between 6 or 7 seconds to receive the bye
I am looking for someone who can make the logo I drew (circeld in blue) into a digital form. the logo does not have to be finished/perfect but I want to see what your skillset is through example UPDATE: it is a close poll between the cursive high star rated one and and @raheel25nov versions. I uploadet som...skillset is through example UPDATE: it is a close poll between the cursive high star rated one and and @raheel25nov versions. I uploadet some more scetches inspired by @raheel. Not sure if they will look good in digital form UPDATE 8/1/21 The new logo from the new scetches found great admiration. Along to the more playful, cursive handwriting looking like. Is there a chance to have tha L like a tree trunk and ending in the circle of the i with the plant.. similar to the 3rd pict...
Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Option to add delete sound files. 6. Branding Thanks Vidhya Sagar Dixit
...the real drawings as faithfully as possible and being detailed with the customization. At the beginning the drawings may take longer because they are made from 0, but when they have been repeated many times, you can take advantage of the bodies and simply change the face and add details such as a tattoo, for example. The deliverable we need is in photoshop with separate layers (background, head, trunk, arms, legs, etc.). 2 - And on the other hand we also need a content creator in social networks, both for posts, reels, tiktok, stories ... This would generally consist of doing the same as the above but drawing celebrities, memes and trends, but all with the style of our drawings. We would define what to do and we would send it to you, it wouldn’t be something you should th...
Hello, I am using Issabel 4 with asterisk 16. At the moment I can only upload audio file at 8000 hz . I would like to know if it's possible to use .wav files at an higher rates like 16000 hz or 48000 hz in order to increase audio quality. Audio files should be used for all the audio parts: IVR, Announcements etc...
Looking for someone experienced in mainly Mirta PBX or Asterisk by itself is less preferred but might be an option.. you would likely be a voice Engineer and will need to train people in use of and maintenance the PBX.
Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Branding Thanks Vidhya Sagar Dixit
The project involves creating an outbound dialer software with IVR functionality.: • Develop a system that is capable of making bulk calls to recipients. • The system must be able to be hosted online and be able to connect to operator E1/SIP Trunk. • The system must be able to schedule phone calls. The system must have a phone book, that includes name, number and sex (male or female) fields, the most important field is the number. The ability to create a group should also be available. • The system should be able to create different Campaigns (Campaign creation). • The system should be able to create IVR Menu’s for specific campaigns. Example Campaign: Training people about financial literacy in rural areas, the course has 6 lessons which wil...
Dear all, I have CISCO router act as a FXO gateway for Asterisk server, it was working find upto the point where the incoming calls are not being ended when hangup , it keep active which make the line always busy,
I need to build an Android app that will receive a list of residents from and API. when pressed on one, a phone call using FreePBX will be initiated. The resident will be able to speak to the user. If the residents presses #0 on his phone, a string will be sent through socket communication to the door opener, and the door will open. Next phase will be an app that will allow Sip Video Calls.
We are looking for someone to run pen tests using sipvicous pentest tool running through python making use of the full tool kit to the best of their ability. The contract will be ongoing and paid at a healthy rate. If you feel you are qualified for this task please get in touch. Thankyou
Need to create a super scaleable conference bridge using an open source PBX. The user management and auth will be handled separately. Knowledge of Asterisk or FreeSwitch is relevant.
I have an ISO image (operating system for Boom Infotainment system for Harley Davidson). I'm interested in unpacking the ISO so that it has intelligible documents that I can edit graphically. Perhaps changing the logo is all I will do. I don't know if this is possible and if not please let me know. I don't know what language it's coded in. The IS...image (operating system for Boom Infotainment system for Harley Davidson). I'm interested in unpacking the ISO so that it has intelligible documents that I can edit graphically. Perhaps changing the logo is all I will do. I don't know if this is possible and if not please let me know. I don't know what language it's coded in. The ISO image is here:
I need you to develope multi tenant pbx, USA for me. must be experienced person
I need you to build Sip trunk pbx for multi tenants call in USA. Person must be experienced in management of trunk and server.
Hi Ameer Ullah A., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I want you setup my sip trunk to fusion pbx and outbount and a inbound rule. I already have extensions
Hi, We have an application which is React JS front end and Django backend. We are looking to incorporate webrtc within this application: We would like 5 features if possible: 1. webrtc - audio to other users on the application 2. webrtc - video to other users on the application 3. webrtc - chat to other users on the application 4. webrtc - softphone that connects to an Asterisk PBX - to click to dial from data on the user panel - basic softphone functionality 5. An autodialer - basically - it loops through the users contacts - dials the number, if it answers, it opens up the screen of the client. The two repos are on github, we have a development server also. Certain user information is on the django backend which the reactJS front end will need up access/update etc.
My name is Dayne Grant. I own a business that uses a CNC router. I want someone to design a 3D relief model of a tree trunk with four independent branches, similar to what is shown in the photo attached.
SIP mobile dialer With LOGIN and history and information
...request. Multitenant. User login with e-mailaddress Admin login, reseller login, tenant admin login, user login. BLF and speeddial buttons, button templates, server templates and tenant_policy Call history Orders and licenses to tenants. Based on MySQL database. To be more clear: we don't use a third party API like Twilio, but the websocket connection is between the clients browser and i.e. an Asterisk server. We have both a prototype of the management interface and the Web dialer interface. Please ask....
goautodial using ASTPP trunk will having problem for autodial. customer answered the call but get silence and it wont pass to agent softphone. within 5 second will drop the call. tested using manual call with no problem with RTP voice complete. i has been try to use other supplier trunk and getting not issue with manual and auto dial from campaign.
Hi guy's, I have a sip provider, but I don't want to install a softphone on my computer, instead I want to use a HTML SIP / JS so I can make a module to use with my CRM. I have find those two references page and it seem it can be acheive. and For this projet, I want to be able to connect with my phone provider and receive a call I can anwser or make a call. Basic HTML5 will do it. Ping me if you have question, we talk about price in the chat.
...and everything else related to keeping the business running. A few months ago I purchased a GoIP4 SIP GSM Gateway, but been having a hard time configuring it, so it hasn't worked as expected up to now. I've got 4 local sim cards in the machine already but I never got them to work properly. I directed the calls to an SIP address on one of my virtual PBX, I want those sim cards to receive and make outgoing calls and I want all 4 people to be able to do so right from their stations at home even though we are not in the same network. At this point, I can receive calls but calls always get disconnected only after 10 to 15 seconds for unknown reasons. When I make calls, the calls are charged to my SIP provider instead of being charged to the SIM card, so I�...
when incoming and outgoing call going in asterisk some time audio not coming
GOIP brand 8 port gsm gateway. Need to configure the equipment to achieve the following: using an app of SIP soft phone, to be able to receive calls from the sim cards in the GOIP GW physically present. One app that can have all the connection for all the phone numbers of the sims in GOIP. Able to make outgoing calls as well, selecting which line/sim for outgoing call. Receive and send SMS texts for each sim as well. Essentially using the GOIP GW, but have access to all sim cards in an app on one mobile device.
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