i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]
...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...
...files securely through https or SMTP. We started the project and have created code that initializes the SD Card and the WiFi access. This functionality was created from the wlan station example and the sdhost_fatfs example. We have determined that to create a secure connection the firmware will require the use of an SSL private key file. We have
...installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer &bu...
I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.
...looking for an image that is sharp with the design bearing a take on 'circles'. I appreciate the use of color. The name should be incorporated within the logo. Please do not use freeware logo sites as they will be rejected and we want unique entries !! The Circles is a platform set up to introduce gatherings (Circles) of knowledge and discussion that help
I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source
...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www
Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples
We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.
I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with
I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else
I require a voicemail drop ...list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.
we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion
Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)
...Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice or
...router have connected a VOIP DEVICE i want to send call THIS VOIP DEVICE now i have 2options 1) Install VPN on server and openwrt ROUTER and bridge network and set a static IP same serial on VOIP DEVICE useing MASQUERADE 2) make remote connections and USE 2nd IP on my voip device i need a solutions to pass call server to local VOIP device i cant use any
...the GSM gateway &3 (SPA3000) Cisco VoIP connected to Internet over a VPN Gateway * This granted GSM & Cisco gateways to see the server in Germany * Server in Germany connected to the VPN gateway via RASPBERRY B3 computer that is connected to the same switch with all VoiP devices. We expect to configure the above VOIP devices with the xCALLY for both
i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not
...expert to fulfill the following: 1- Write a script for VICIDIAL installation with a single command line from [login to view URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested
Hi. We are developing a mobile app connected to a web server. It is a freeware app for animal rescuers. We expect thousands of users to use the application and it appears that google charges $ 4 per 1000 loads for the location details beyond its free usage of $ 200. We can not afford that since each mission will involved marking 2 locations or more
...5" (freeware). You have to install scribus and afterwards paste side by side the text from a pdf into scribus (see pictures). The layout is already done, but text consists of curves at the moment and cannot be edited therefore. Finally I need five updated scribus files. The job cannot be done with illustrator, only with scribus 1.5.X (freeware for
A- 1- Want VOIP app to be working in the background even after closing the app. 2- Auto restart app after phone rebooted. B- Upload to google play. C- Give me source code after finish the job.
I have an open Freelance HR/Recruiter position to help close 1-3 positions for the International Telecommunication Company based in NY The experience in hiring in Telecommunications, IT, Sales (preferable) Price is negotiable
USA Nationwide Business VoIP Company looking for business leads. We offer Cloud Based Telephone Systems and Hosted IVR Solutions for BUSINESSES ONLY. Our "sweet spot" is the 5-25 telephone deals. Larger companies are good too but not a primary [login to view URL] only operate in the lower 48 United States. ALL Leads must have Business Name, Address, Decision
I need my gateway to be set up so that I can receive international voip traffic and forwarded using the simcards on the gateway. You set the route and Asterisk server on VPS. This should take just few minutes
we have a task allocation web system and want to integrate ring central VOIP API so call centre agents can receive calls on our web app dashboard and allocate tasks easily, we need you to manage queues and simultaneously calling so all agents will keep busy to serve more customer calls Only experts are required who have already done this kind of work
Need a browser extension or application for both windows and mac. The soft phone will require an agent to log in and be able to show when an agent is logged in, we also need to understand when the agent is available, on do not disturb or on a call. Need the availability to move a call from one agent to another agent. All call details will need to
...interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation from development; - log from customers and admins about changes; - each user have your color/logo (size from picture
Phase 1 Need to setup asteriskNow with ippbx (tata telecom). Setting up sip trunk. redirecting calls on specific numbers to specific sip no. At specific work hours or if specified through API, then to redirect calls to Mobile no. Setup recording of calls.
...using Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have
I have a very simple job to do. Altogether 781 pages have to be modivied with dtp-software scribus 1.5 (freeware). You have to install scribus and afterwards paste side by side the text from a pdf into scribus (see pictures). The jpgs show how to do it. The pdf shows one of the five files. You can copy the text from there. The 781 pages consist of
I need the Linphone client rebranded for Android/IOS and Windows/OSX. Simply with our logo / app-icon, removing options for creating or using linphone account, so only sip account is configurable. Furthermore changed to our colors. Packages for IOS and Android should be ready for deployment to the public stores, so we need the sourcecode when
Datalink International is seeking a skilled Android and iOS Application developer. Must have 5 + years of proven development skills. All A...developer. Must have 5 + years of proven development skills. All Apps developed will link over GSM Cellular networks to existing GPS and SMS Host Software. Must be familiar with SIP and serial data connections
...that allow me to send calls to the mobile via SIP ( VoIP ) and this calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via ...
I have an extremely small (8 phone) 3CX VOIP system in my business and need some help sorting a few issues out that I cannot seem to fix. I need someone who is 3CX experienced, as I have used general VOIP people off here before but its never perfectly right, so prefer someone with in depth 3CX knowledge rather than an all rounder.
I want an Android which has a few calculators - Loan EMI Fixed Deposits Recurring Deposits Direct MF / SIP Simple Interest Compound Interest Target version must be Android 8 Oreo and minimum supported version should be Lollipop. Design of app should be good.
...create everyday a new CDR Table i need a CDR page where i can check all the calls report like cdr by account name/gateway name or ip/client name or ip/ make sure i can use a SIP account as a routing gateway I want to make same to same VOS3000 softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work